Bibliography

1

M. Abe and J. O. Smith, ``Design criteria for simple sinusoidal parameter estimation based on quadratic interpolation of FFT magnitude peaks,'' Audio Engineering Society Convention, San Francisco, 2004,
Preprint 6256.

2

J. Abel, ``private communication,'' 2006.

3

J. S. Abel, ``Expressions relating frequency, critical-band rate, and critical bandwidth,'' 1997,
submitted for publication.

4

E. Aboutanios and B. Mulgrew, ``Iterative frequency estimation by interpolation on fourier coefficients,'' IEEE Transactions on Signal Processing, vol. 53, pp. 1237-1242, Apr. 2005.

5

M. Abramowitz and I. A. Stegun, eds., Handbook of Mathematical Functions,
New York: Dover, 1965.

6

M. Ali, Adaptive Signal Representation with Application in Audio Coding,
PhD thesis, University of Minnesota, Mar. 1996.

7

J. B. Allen, ``Short term spectral analysis, synthesis, and modification by discrete Fourier transform,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-25, pp. 235-238, June 1977.

8

J. B. Allen, ``Application of the short-time Fourier transform to speech processing and spectral analysis,'' Proc. IEEE ICASSP-82, pp. 1012-1015, 1982.

9

J. B. Allen and L. R. Rabiner, ``A unified approach to short-time Fourier analysis and synthesis,'' Proc. IEEE, vol. 65, pp. 1558-1564, Nov. 1977.

10

X. Amatriain, J. Bonada, A. Loscos, and X. Serra, ``Spectral processing,'' in DAFX - Digital Audio Effects (U. Zölzer, ed.), pp. 373-438, West Sussex, England: John Wiley and Sons, LTD, 2002,
http://www.dafx.de/.

11

B. S. Atal and L. S. Hanauer, ``Speech analysis and synthesis by linear prediction of the speech wave,'' Journal of the Acoustical Society of America, vol. 50, pp. 637-655, 1971.

12

F. Auger and P. Flandrin, ``Improving the readibility of time-frequency and time-scale representations by the reassignment method,'' IEEE Transactions on Signal Processing, vol. 43, pp. 1068-1089, May 1995.

13

L. C. Barbosa, ``A maximum-energy-concentration spectral window,'' IBM Journal of Research and Development, vol. 30, pp. 321-325, May 1986.

14

M. Bellanger, ``Improved design of long FIR filters using the frequency masking technique,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Atlanta, (New York), IEEE Press, May 1996,
Paper DSP1.1.

15

L. L. Beranek, Acoustics,
http://asa.aip.org/publications.html: American Institute of Physics, for the Acoustical Society of America, 1986,
1st ed. 1954.

16

M. Bosi and R. E. Goldberg, Introduction to Digital Audio Coding and Standards,
Boston: Kluwer Academic Publishers, 2003.

17

M. Bosi, K. Brandenburg, S. Quackenbush, L. Fielder, K. Akagin, H. Fuchs, M. Dietz, J. Herre, G. Davidson, and Y. Oikawa, ``ISO / IEC MPEG-2 advanced audio coding,'' Audio Engineering Society Convention, vol. Preprint 4382, Nov. 1996,
36 pages. See also ISO/IEC International Standard IS 13818-7 entitled ``MPEG-2 Advanced Audio Coding,'' April, 1997.

18

L. Bosse, ``An experimental high fidelity perceptual audio coder,'' tech. rep., Elec. Engineering Dept., Stanford University (CCRMA), Mar. 1998,
Music 420 Project Report, https://ccrma.stanford.edu/~jos/bosse/.

19

R. C. Boulanger, ed., The Csound Book: Perspectives in Software Synthesis, Sound Design, Signal Processing, and Programming,
MIT Press, Mar. 2000.

20

J. Bouvrie and T. Ezzat, ``An incremental algorithm for signal reconstruction from short-time fourier transform magnitude,'' in INTERSPEECH 2006 -- ICSLP, 2006,
http://cbcl.mit.edu/publications/ps/signalrec_ICSLP06.pdf.

21

G. E. P. Box and G. M. Jenkins, Time Series Analysis,
San Francisco: Holden Day, 1976.

22

S. Boyd and L. Vandenberghe, Convex Optimization,
Cambridge University Press, Feb. 2004,
http://www.stanford.edu/~boyd/cvxbook/.

23

R. Bracewell, The Fourier Transform and its Applications,
New York: McGraw-Hill, 1965.

24

K. Brandenburg, ``Perceptual coding of high quality digital audio,'' in Applications of DSP to Audio & Acoustics (M. Kahrs and K. Brandenburg, eds.), pp. 39-83, Boston/Dordrecht/London: Kluwer Academic Publishers, 1998.

25

K. Brandenburg and M. Bosi, ``Overview of MPEG audio: Current and future standards for low-bit-rate audio coding,'' Journal of the Audio Engineering Society, vol. 45, pp. 4-21, Jan./Feb. 1997.

26

A. S. Bregman, Auditory Scene Analysis: the Perceptual Organization of Sound,
Cambridge, MA: MIT Press, 1990.

27

R. Bristow-Johnson, ``Wavetable synthesis 101, a fundamental perspective,'' presented at 101st AES Convention, no. Preprint 4400, 1996,
http://www.musicdsp.org/files/Wavetable-101.pdf.

28

R. Bristow-Johnson, ``Tutorial on floating-point versus fixed-point,'' Audio Engineering Society Convention, Oct. 2008.

29

J. C. Brown, ``Calculation of a constant Q spectral transform,'' Journal of the Acoustical Society of America, vol. 89, no. 1, pp. 425-434, 1991.

30

J. C. Brown and M. S. Puckette, ``An efficient algorithm for the calculation of a constant Q transform,'' Journal of the Acoustical Society of America, vol. 92, no. 5, pp. 2698-2701, 1992.

31

T. Brown and M. M. Wang, ``An iterative algorithm for single-frequency estimation,'' IEEE Transactions on Signal Processing, vol. 50, no. 11, pp. 2671-2682, 1993.

32

C. S. Burrus, ``Chebyshev or equal ripple error approximation filters,'' tech. rep., Connexions, Nov. 2008,
http://cnx.org/content/m16895/latest/.

33

R. J. Cassidy and J. O. Smith III, ``Efficient time-varying loudness estimation via the hopping Goertzel DFT,'' in Proceedings of the IEEE International Midwest Symposium on Circuits and Systems (MWSCAS-2007), Aug. 2007.

34

C. Chafe, D. Jaffe, K. Kashima, B. Mont-Reynaud, and J. O. Smith, ``Techniques for note identification in polyphonic music,'' in Proceedings of the 1985 International Computer Music Conference, Vancouver, Computer Music Association, searchable at http://quod.lib.umich.edu/i/icmc/, 1985.

35

H. Chamberlin, Musical Applications of Microprocessors,
New Jersey: Hayden Book Co., Inc., 1980.

36

D. C. Champeney, A Handbook of Fourier Theorems,
Cambridge University Press, 1987.

37

D. G. Childers, ed., Modern Spectrum Analysis,
New York: IEEE Press, 1978.

38

J. M. Chowning, ``The synthesis of complex audio spectra by means of frequency modulation,'' Journal of the Audio Engineering Society, vol. 21, no. 7, pp. 526-534, 1973,
reprinted in [236].

39

J. M. Chowning, ``Frequency modulation synthesis of the singing voice,'' in Current Directions in Computer Music Research (M. V. Mathews and J. R. Pierce, eds.), pp. 57-63, Cambridge, MA: MIT Press, 1989.

40

J. M. Chowning, ``private communication,'' 2006.

41

J. M. Chowning and D. Bristow, FM Theory and Applications,
Tokyo: Yamaha Corporation, 1986.

42

R. V. Churchill, Complex Variables and Applications,
New York: McGraw-Hill, 1960.

43

L. Cohen, Time-Frequency Analysis,
Englewood Cliffs, NJ: Prentice-Hall, 1995.

44

B. Conolly and I. J. Good, ``A table of discrete Fourier transform pairs,'' SIAM J. Applied Math, vol. 32, no. 4, pp. 810-822, 1977.

45

P. Cook and G. Scavone, Synthesis Tool Kit in C++, Version 4,
https://ccrma.stanford.edu/software/stk/, 2010,
see also https://ccrma.stanford.edu/~jos/stkintro/.

46

P. R. Cook, Identification of Control Parameters in an Articulatory Vocal Tract Model, with Applications to the Synthesis of Singing,
PhD thesis, Elec. Engineering Dept., Stanford University (CCRMA), Dec. 1990,
http: //www.cs.princeton.edu/~prc/.

47

P. R. Cook, Real Sound Synthesis for Interactive Applications,
A. K. Peters, L.T.D., 2002.

48

T. Cover and J. Thomas, Elements of Information Theory,
New York: John Wiley and Sons, Inc., 1991.

49

R. Crochiere, ``A weighted overlap-add method of short-time Fourier analysis/synthesis,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-28, pp. 99-102, Feb 1980.

50

R. Crochiere and L. R. Rabiner, Multirate Digital Signal Processing,
Englewood Cliffs, NJ: Prentice-Hall, 1983.

51

A. Croisiere, D. Esteban, and C. Galand, ``Perfect channel splitting by use of interpolation/decimation/tree decomposition techniques,'' in Proc. Int. Symp. Info., Circuits and Systems, Patras, Greece, 1976.

52

O. Darrigol, ``The acoustic origins of harmonic analysis,'' Archive for History of the Exact Sciences, vol. 61, July 2007.

53

G. Davidson, L. Fielder, and M. Antill, ``Low-complexity transform coder for satellite link applications,'' Audio Engineering Society Convention, vol. Preprint 2966, no. Session-Paper no. F-I-6, pp. 1-22, 1990.

54

A. de Cheveigné, ``Pitch perception models -- a historical review,'' in Proceedings of the 18th International Conference on Acoustics (ICA-04), Kyoto, Japan, p. Tu2.H.2, 2004,
http://recherche.ircam.fr/equipes/pcm/cheveign/pss/2004_ICA.pdf.

55

A. de Cheveigne and H. Kawahara, ``YIN, a fundamental frequency estimator for speech and music,'' Journal of the Acoustical Society of America, vol. 111, no. 4, pp. 1917-1930, 2002.

56

J. R. Deller Jr., J. G. Proakis, and J. H. Hansen, Discrete-Time Processing of Speech Signals,
New York: Macmillan, 1993.

57

J. R. Deller Jr., J. H. L. Hansen, and J. G. Proakis, Discrete-Time Processing of Speech Signals,
New York: John Wiley and Sons, Inc., Mar. 2001.

58

P. Depalle and T. Hélie, ``Extraction of spectral peak parameters using a short-time Fourier-transform modeling and no-sidelobe windows,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 1997.

59

P. A. M. Dirac, The Principles of Quantum Mechanics, Fourth Edition,
New York: Oxford University Press, 1958-2001.

60

C. Dodge and T. A. Jerse, Computer Music: Synthesis, Composition, and Performance,
New York: Wadsworth Publication Co., 1997.

61

C. L. Dolph, ``A current distribution for broadside arrays which optimizes the relationship between beam width and side-lobe level,'' Proceedings of the IRE, vol. 34, pp. 335-348, 1946.

62

M. Dolson, ``The phase vocoder: A tutorial,'' Computer Music Journal, vol. 10, no. 4, pp. 14-27, 1986.

63

S. Dostrovsky, ``Early vibration theory: Physics and music in the seventeenth century,'' Archive for History of the Exact Sciences, vol. 14, no. 3, 1975.

64

B. Doval and X. Rodet, ``Estimation of fundamental frequency of musical sound signals,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Toronto, (New York), pp. 3657-3660, vol. 5, IEEE Press, April 14-17, 1991.

65

B. Doval and X. Rodet, ``Fundamental frequency estimation using a new harmonic matching method,'' in Proceedings of the 1991 International Computer Music Conference, Montréal, pp. 555-558, Computer Music Association, searchable at http://quod.lib.umich.edu/i/icmc/, 1991.

66

DSP Committee, ed., Programs for Digital Signal Processing,
New York: IEEE Press, 1979.

67

DSP Committee, ed., Programs for Digital Signal Processing II,
New York: IEEE Press, 1979.

68

H. W. Dudley, ``The vocoder,'' Bell Labs Rec., vol. 18, pp. 122-126, 1939,
Reprinted in [243, pp. 347-351].

69

B. Eaglestone and S. Oates, ``Analytical tools for group additive synthesis,'' in Proceedings of the 1990 International Computer Music Conference, Glasgow, Computer Music Association, searchable at http://quod.lib.umich.edu/i/icmc/, 1990.

70

D. P. W. Ellis, ``A phase vocoder in Matlab,'' 2002,
Web resource.

71

H. Fastl and E. Zwicker, Psychoacoustics: Facts and Models,
Berlin: Springer Verlag, 2006,
third edition, 462pp., CD-ROM.

72

J. A. Fessler and B. P. Sutton, ``Nonuniform fast Fourier transforms using min-max interpolation,'' IEEE Transactions on Signal Processing, vol. 51, pp. 560-574, Feb. 2003.

73

K. Fitz and L. Haken, ``On the use of time-frequency reassignment in additive sound modeling,'' Journal of the Audio Engineering Society, vol. 50, pp. 879-893, Nov. 2002.

74

J. L. Flanagan, Speech Analysis, Synthesis, and Perception,
New York: Springer Verlag, 1972.

75

J. L. Flanagan, ``Voices of men and machines,'' Journal of the Acoustical Society of America, vol. 51, pp. 1375-1387, Mar. 1972,
Reprinted in [243, pp. 4-16].

76

J. L. Flanagan and R. M. Golden, ``Phase vocoder,'' Bell System Technical Journal, vol. 45, pp. 1493-1509, Nov. 1966,
Reprinted in [243, pp. 388-404].

77

N. H. Fletcher and T. D. Rossing, The Physics of Musical Instruments, 2nd Edition,
New York: Springer Verlag, 1998.

78

A. Franck, Efficient Algorithms for Arbitrary Sample Rate Conversion with Application to Wave Field Synthesis,
PhD thesis, Technischen Universität Ilmeneau, Nov. 2011.

79

D. K. Frederick and A. B. Carlson, Linear Systems in Communication and Control,
New York: John Wiley and Sons, Inc., 1971.

80

M. Frigo and S. G. Johnson, ``FFTW: An adaptive software architecture for the FFT,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Seattle, vol. 3, (New York), pp. 1381-1384, IEEE Press, 1998,
http://www.fftw.org/.

81

S. A. Fulop and K. Fitz, ``A spectrogram for the twenty-first century,'' Acoustics Today, pp. 26-33, July 2006.

82

T. J. Gardner and M. O. Magnesco, ``Sparse time-frequency representations,'' Proc. National Academy of Sciences of the United States of America, vol. 103, April 2006.

83

E. B. George and M. J. T. Smith, ``A new speech coding model based on least-squares sinusoidal representation,'' Proc. IEEE ICASSP-87, vol. 3, pp. 1641-1644, 1987.

84

E. B. George and M. J. T. Smith, ``Analysis-by-synthesis/Overlap-add sinusoidal modeling applied to the analysis and synthesis of musical tones,'' Journal of the Audio Engineering Society, vol. 40, no. 6, pp. 497-516, 1992.

85

A. Gerzso, ``Density of spectral components: Preliminary experiments,'' tech. rep., IRCAM, 1978,
Report no. 31/80, abstract and ordering information on-line at http://mediatheque.ircam.fr/articles/index-e.html.

86

P. E. Gill, W. Murray, and M. H. Wright, Practical Optimization,
New York: Academic Press, 1981.

87

B. R. Glasberg and B. C. J. Moore, ``Derivation of auditory filter shapes from notched-noise data,'' Hearing Research, vol. 47, pp. 103-138, 1990.

88

B. R. Glasberg and B. C. J. Moore, ``A model of loudness applicable to time-varying sounds,'' Journal of the Audio Engineering Society, vol. 50, pp. 331-342, May 2002.

89

S. Golden, ``Exponential polynomial signals: Estimation, analysis, and applications,'' 1995,
citeseer.nj.nec.com/golden95exponential.html.

90

S. Golden and B. Friedlander, ``Estimation and statistical analysis of exponential polynomial signals.''
citeseer.nj.nec.com/golden98estimation.html.

91

S. Golden and B. Friedlander, ``Maximum likelihood estimation, analysis, and applications of exponential polynomial signals.''
citeseer.nj.nec.com/golden99maximum.html.

92

G. H. Golub and C. F. Van Loan, Matrix Computations, 3rd Edition,
Baltimore: The Johns Hopkins University Press, 1996.

93

M. Goodwin and A. Kogon, ``Overlap-add synthesis of nonstationary sinusoids,'' in Proceedings of the 1995 International Computer Music Conference, Banff, Computer Music Association, searchable at http://quod.lib.umich.edu/i/icmc/, 1995,
citeseer.nj.nec.com/goodwin95overlapadd.html.

94

M. Goodwin and A. Kogon, ``Overlap-add synthesis of nonstationary sinusoids,'' in International Computer Music Conference, 1995.

95

R. M. Gray and L. D. Davisson, An Introduction to Statistical Signal Processing,
Cambridge University Press, 2003,
http://www-ee.stanford.edu/~gray/sp.pdf.

96

D. D. Greenwood, ``A cochlear frequency-position function for several species--29 years later,'' Journal of the Acoustical Society of America, vol. 87, pp. 2592-605, June 1990.

97

J. Grey, An Exploration of Musical Timbre,
PhD thesis, Stanford University, Feb. 1975,
available as CCRMA Technical Report STAN-M-2, Music Dept., Stanford University.

98

D. W. Griffin and J. S. Lim, ``Signal estimation from modified short-time Fourier transform,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-32, pp. 236-243, Apr 1984.

99

D. W. Griffin and J. S. Lim, ``Multiband-excitation vocoder,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-36, pp. 236-243, Feb 1988.

100

D. Halliday, R. Resnick, and J. Walker, Extended, Fundamentals of Physics, 6th Edition,
New York: John Wiley and Sons, Inc., 2000.

101

F. J. Harris, ``On the use of windows for harmonic analysis with the discrete Fourier transform,'' Proceedings of the IEEE, vol. 66, pp. 51-83, Jan 1978.

102

W. Hartmann, Signals, Sound, and Sensation,
New York: AIP Press, 1997,
647 pp., 221 illustrations, hardcover.

103

M. H. Hayes, J. S. Lim, and A. V. Oppenheim, ``Signal reconstruction from phase or magnitude,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-28, pp. 672-680, Dec 1980.

104

D. Hejna and B. R. Musicus, ``The solafs time-scale modification algorithm,'' tech. rep., BBN Technical Report, July 1991,
http://www.ee.columbia.edu/ dpwe/papers/HejMus91-solafs.pdf.

105

H. D. Helms, ``Digital filters with equiripple or minimax responses,'' IEEE Transactions on Audio and Electroacoustics, vol. 19, pp. 87-94, Mar. 1971.

106

W. J. Hess, Algorithms and Devices for Pitch Determination of Speech-Signals,
Berlin: Springer-Verlag, 1983.

107

A. Horner, J. Beauchamp, and L. Haken, ``Methods for multiple wavetable synthesis of musical instrument tones,'' Journal of the Audio Engineering Society, vol. 41, pp. 336-356, May 1993.

108

P. Huang, Artificial Reverberation using the Digital Waveguide Mesh,
PhD thesis, Music Department, Stanford University,
in preparation.

109

B. Hutchins, Musical Engineer's Handbook,
Ithaca, New York: Electronotes, 1975.

110

T. Irino and H. Kawahara, ``Signal reconstruction from modified auditory wavelet transform,'' IEEE Transactions on Signal Processing, vol. 41, no. 12, pp. 3549-3554, 1993.

111

T. Irino and R. D. Patterson, ``A time-domain, level-dependent auditory filter: The gammachirp,'' Journal of the Acoustical Society of America, vol. 101, pp. 412-419, 1997.

112

T. Irino and M. Unoki, ``A time-varying analysis/synthesis auditory filterbank using the gammachirp,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Seattle, (New York), IEEE Press, 1998.

113

ISE/IEC JTC 1/SC 29/WG 11, ISO/IEC 11172-3: Information Technology - Coding of Moving Pictures and Associated Audio for Digital Storage Media at up to about 1.5 Mbit/s - Part 3: Audio,
Motion Picture Experts Group, 1993.

114

T. Kailath, A. H. Sayed, and B. Hassibi, Linear Estimation,
Englewood Cliffs, NJ: Prentice-Hall, Apr. 2000.

115

J. F. Kaiser, ``Using the ${I}_0-\sinh$ window function,'' IEEE Transactions on Circuits and Systems--I: Fundamental Theory and Applications, pp. 20-23, April 22-25 1974,
Reprinted in [67], pp. 123-126.

116

L. J. Karam and J. H. McClellan, ``Complex Chebyshev approximation for FIR filter design,'' IEEE Transactions on Circuits and Systems--II: Analog and Digital Signal Processing, vol. 42, no. 3, pp. 207-216, 1995.

117

L. J. Karam and J. H. McClellan, ``Design of optimal digital FIR filters with arbitrary magnitude and phase responses,'' in ISCAS-96, 1996,
http://www.eas.asu.edu/~karam/papers/iscas96_km.html.

118

M. Karjalainen, ``A new auditory model for the evaluation of sound quality of audio systems,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Tampa, Florida, (New York), pp. 608-611, IEEE Press, 1985.

119

M. Karjalainen and J. O. Smith, ``Body modeling techniques for string instrument synthesis,'' in Proceedings of the 1996 International Computer Music Conference, Hong Kong, pp. 232-239, Computer Music Association, searchable at http://quod.lib.umich.edu/i/icmc/, Aug. 1996.

120

S. M. Kay, Modern Spectral Estimation,
Englewood Cliffs, NJ: Prentice-Hall, 1988.

121

S. M. Kay, Fundamentals of Statistical Signal Processing, Volume I: Estimation Theory,
Englewood Cliffs, NJ: Prentice-Hall, 1993.

122

S. Kay, ``A fast and accurate single frequency estimator,'' IEEE Trans. Acoustics, Speech, and Signal Processing, vol. 37, pp. 1987-1990, Dec. 1989.

123

Y. Kim and J. O. Smith, ``A speech feature based on bark frequency warping - the non-uniform linear prediction (nlp) cepstrum,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 1999.

124

A. Klapuri, ``Multipitch estimation and sound separation by the spectral smoothness principle,'' in IEEE International Conference on Acoustics, Speech, and Signal Processing, (Salt Lake City, Utah, USA), May 2001.

125

A. Klapuri, ``Automatic transcription of music,'' in Proceedings of the Stockholm Musical Acoustics Conference (SMAC-03), http://www.speech.kth.se/smac03/, (Stockholm), Royal Swedish Academy of Music, Aug. 2003,
also http://www.cs.tut.fi/~klap/iiro/index.html.

126

A. Klapuri, ``Multiple fundamental frequency estimation by harmonicity and spectral smoothness,'' IEEE Transactions on Speech and Audio Processing, vol. 11, no. 6, pp. 804-816, 2003.

127

A. Klapuri, ``A perceptually motivated multiple-F0 estimation method,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 2005,
http://www.cs.tut.fi/sgn/arg/klap/waspaa2005.pdf.

128

D. Klatt, ``Software for a cascade/parallel formant synthesizer,'' Journal of the Acoustical Society of America, vol. 67, pp. 13-33, 1980.

129

D. Klatt, ``Review of text-to-speech conversion for english,'' Journal of the Acoustical Society of America, vol. 82, pp. 737-793, Sept. 1987.

130

P. Kleczkowski, ``Group additive synthesis,'' Computer Music Journal, vol. 13, no. 1, pp. 12-20, 1989.

131

D. E. Knuth, The Art of Computer Programming, Volumes 1-3,
Reading MA: Addison-Wesley, 1998,
First published in 1973.

132

W. Koenig, H. K. Dunn, and L. Y. Lacy, ``The sound spectrograph,'' Journal of the Acoustical Society of America, vol. 18, no. 1, pp. 19-49, 1946.

133

M. Z. Komodromos, S. F. Russel, and P. T. P. Tang, ``Design of FIR Hilbert transformers and differentiators in the complex domain,'' IEEE Transactions on Circuits and Systems--II: Analog and Digital Signal Processing, vol. 45, pp. 64-67, Jan 1998.

134

P. J. Kootsookos and R. C. Williamson, ``FIR approximation of fractional sample delay systems,'' IEEE Transactions on Circuits and Systems--II: Analog and Digital Signal Processing, vol. 43, pp. 40-48, Feb 1996.

135

T. I. Laakso, V. Välimäki, M. Karjalainen, and U. K. Laine, ``Splitting the Unit Delay--Tools for Fractional Delay Filter Design,'' IEEE Signal Processing Magazine, vol. 13, pp. 30-60, Jan. 1996.

136

H. Landau and H. Pollak, ``Prolate-spheroidal wave functions, Fourier analysis and uncertainty--ii,'' Bell System Technical Journal, vol. 40, pp. 65-84, Jan. 1961.

137

J. Laroche, ``The use of the matrix pencil method for the spectrum analysis of musical signals,'' Journal of the Acoustical Society of America, vol. 94, pp. 1958-1965, Oct. 1993.

138

J. Laroche, ``Time and pitch scale modification of audio signals,'' in Applications of DSP to Audio & Acoustics (M. Kahrs and K. Brandenburg, eds.), pp. 279-309, Boston/Dordrecht/London: Kluwer Academic Publishers, 1998.

139

J. Laroche, ``Synthesis of sinusoids via non-overlapping inverse fourier transform,'' IEEE Transactions on Speech and Audio Processing, vol. 8, pp. 471-477, July 2000.

140

J. Laroche and M. Dolson, ``About this phasiness business,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 1997.

141

J. Laroche and M. Dolson, ``Improved phase vocoder time-scale modification of audio,'' IEEE Transactions on Speech and Audio Processing, vol. 7, no. 3, pp. 323-32, 1999.

142

J. Laroche and M. Dolson, ``New phase-vocoder techniques for pitch-shifting, harmonizing, and other exotic effects,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), pp. 91-94, IEEE Press, Oct. 17-20, 1999.

143

J. Laroche and M. Dolson, ``New phase-vocoder techniques for real-time pitch shifting, chorusing, harmonizing, and other exotic audio modifications,'' Journal of the Audio Engineering Society, vol. 47, pp. 928-936, Nov. 1999.

144

S. N. Levine and J. O. Smith, ``A sines+transients+noise audio representation for data compression and time/pitch-scale modifications,'' Audio Engineering Society Convention, vol. Session on Analysis and Synthesis of Sound, no. preprint 4781, 1998,
https://ccrma.stanford.edu/~scottl/papers.html.

145

S. N. Levine and J. O. Smith, ``Improvements to the switched parametric and transform audio coder,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 1999.

146

S. N. Levine and J. O. Smith, ``A compact and malleable sines+transients+noise model for sound,'' in Analysis, Synthesis, and Perception of Musical Sounds: The Sound of Music (J. Beauchamp, ed.), Berlin: Springer-Verlag, 2006.

147

S. N. Levine, T. S. Verma, and J. O. Smith, ``Alias-free multiresolution sinusoidal modeling for polyphonic, wideband audio,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 1997,
https://ccrma.stanford.edu/~scottl/papers.html.

148

S. N. Levine, T. S. Verma, and J. O. Smith, ``Multiresolution sinusoidal modeling for wideband audio with modifications,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Seattle, (New York), IEEE Press, 1998,
https://ccrma.stanford.edu/~scottl/papers.html.

149

S. N. Levine, Audio Representations for Data Compression and Compressed Domain Processing,
PhD thesis, Electrical Engineering Dept., Stanford University (CCRMA), Dec. 1998,
https://ccrma.stanford.edu/~scottl/thesis.html.

150

M. J. Lighthill, Introduction to Fourier Analysis,
Cambridge University Press, Jan. 1958.

151

L. Ljung and T. L.Soderstrom, ``The Steiglitz-McBride algorithm revisited--convergence analysis and accuracy aspects,'' IEEE Transactions on Automatic Control, vol. 26, pp. 712-717, June 1981,
See also the function stmcb() in the Matlab Signal Processing Toolbox.

152

L. Ljung and T. L. Soderstrom, Theory and Practice of Recursive Identification,
Cambridge, MA: MIT Press, 1983.

153

M. Lobo, L. Vandenberghe, S. Boyd, and H. Lebret, ``Applications of second-order cone programming,'' Linear Algebra and its Applications, vol. 284, pp. 193-228, Nov. 1998,
special issue on Linear Algebra in Control, Signals and Image Processing. http://www.stanford.edu/~boyd/socp.html.

154

H.-L. Lu, Toward a High-Quality Singing Synthesizer with Vocal Texture Control,
PhD thesis, Elec. Engineering Dept., Stanford University (CCRMA), July 2002,
https://ccrma.stanford.edu/~vickylu/thesis/.

155

P. Lynch, ``The Dolph-Chebyshev window: A simple optimal filter,'' America Meteorological Society Journal of the Online, vol. 125, pp. 655-660, Apr. 1997.

156

R. Lyons, Understanding Digital Signal Processing,
Upper Saddle River, NJ: Prentice Hall PTR, 2001.

157

J. Makhoul, ``Linear prediction: A tutorial review,'' Proceedings of the IEEE, vol. 63, pp. 561-580, Apr. 1975.

158

D. Malah and J. L. Flanagan, ``Frequency scaling of speech signals by transform techniques,'' Bell System Technical Journal, vol. 60, pp. 2107-2156, Nov 1981.

159

H. S. Malvar, ``Lapped transforms for efficient transform/subband coding,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. 38, pp. 969-978, June 1990.

160

H. S. Malvar, Signal Processing with Lapped Transforms,
Boston: Artech House, 1992.

161

H. S. Malvar and D. H. Staelin, ``The lot: Transform coding without blocking effects,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. 37, pp. 553-559, Apr 1989.

162

J. D. Markel and A. H. Gray, Linear Prediction of Speech,
New York: Springer Verlag, 1976.

163

J. Marks and J. Polito, ``Modeling piano tones,'' in Proceedings of the 1986 International Computer Music Conference, The Hague, Computer Music Association, searchable at http://quod.lib.umich.edu/i/icmc/, 1986.

164

J. S. Marques and L. B. Almeida, ``Frequency-varying sinusoidal modeling of speech,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. 37, pp. 763-765, May 1989.

165

D. C. Massie, ``Wavetable sampling synthesis,'' in Applications of DSP to Audio & Acoustics (M. Kahrs and K. Brandenburg, eds.), pp. 311-341, Boston/Dordrecht/London: Kluwer Academic Publishers, 1998.

166

M. V. Mathews, ``The digital computer as a musical instrument,'' Science, vol. 142, no. 11, pp. 553-557, 1963.

167

M. V. Mathews, The Technology of Computer Music,
Cambridge, MA: MIT Press, 1969.

168

I. G. Mattingly, ``Speech synthesis for phonetic and phonological models,'' in Current Trends in Linguistics (T. A. Sebeok, ed.), vol. 12, pp. 2451-2487, The Hague: Mouton, 1974,
http://www.mindspring.com/~ssshp/ssshp_cd/im_home.htm.

169

R. J. McAulay and T. F. Quatieri, ``Speech analysis-synthesis based on a sinusoidal representation,'' Tech. Rep. 693, Lincoln Laboratory, MIT, 1985.

170

R. J. McAulay and T. F. Quatieri, ``Phase coherence in speech reconstruction for enhancement and coding applications,'' Proc. IEEE ICASSP-89, vol. 1, pp. 207-210, 1989.

171

R. J. McAulay and T. F. Quatieri, ``Pitch estimation and voicing detection based on a sinusoidal speech model,'' Proc. IEEE ICASSP-90, vol. 1, pp. 249-252, 1990.

172

R. J. McAulay and T. F. Quatieri, ``Sine-wave phase coding at low data rates,'' Proc. IEEE ICASSP-91, vol. 1, pp. 577-580, 1991.

173

R. J. McAulay and T. F. Quatieri, ``Mid-rate coding based on a sinusoidal representation of speech,'' Proc. IEEE ICASSP-85, vol. 3, pp. 945-948, 1985.

174

R. J. McAulay and T. F. Quatieri, ``Speech analysis/synthesis based on a sinusoidal representation,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-34, pp. 744-754, Aug 1986.

175

J. H. McClellan, R. W. Schafer, and M. A. Yoder, DSP First: A Multimedia Approach,
Englewood Cliffs, NJ: Prentice-Hall, 1998,
Tk5102.M388.

176

J. H. McClellan, T. W. Parks, and L. R. Rabiner, ``A computer program for designing optimum FIR linear phase digital filters,'' IEEE Transactions on Audio and Electroacoustics, vol. 21, pp. 506-526, Dec. 1973,
program also contained in [224].

177

B. C. J. Moore and B. R. Glasberg, ``A revision of Zwicker's loudness model,'' Acta Acustica, vol. 82, pp. 335-345, 1996.

178

B. C. J. Moore, ``Distribution of auditory-filter bandwidths at 2 kHz in young normal listeners,'' Journal of the Acoustical Society of America, vol. 81, pp. 1633-1635, May 1987.

179

B. C. J. Moore, An Introduction to the Psychology of Hearing,
New York: Academic Press, 1997.

180

B. C. J. Moore and B. R. Glasberg, ``Suggested formulae for calculating auditory-filter bandwidths and excitation patterns,'' Journal of the Acoustical Society of America, vol. 74, pp. 750-753, Sept. 1983.

181

B. C. J. Moore, R. W. Peters, and B. R. Glasberg, ``Auditory filter shapes at low center frequencies,'' Journal of the Acoustical Society of America, vol. 88, pp. 132-140, July 1990.

182

B. C. J. Moore, B. R. Glasberg, and T. Baer, ``A model for the prediction of thresholds, loudness, and partial loudness,'' Journal of the Audio Engineering Society, vol. 45, pp. 224-249, Apr. 1997.

183

F. R. Moore, Elements of Computer Music,
Englewood Cliffs, NJ: Prentice Hall, 1990.

184

J. A. Moorer, ``The hetrodyne filter as a tool for analysis of transient waveforms,'' Tech. Rep. AIM-208, Stanford Artificial Intelligence Laboratory, Computer Schience Dept., Stanford University, July 1973.

185

J. A. Moorer, On the Segmentation and Analysis of Continuous Musical Sound by Digital Computer,
PhD thesis, Computer Science Dept., Stanford University, 1975,
CCRMA Technical ReportSTAN-M-3.

186

J. A. Moorer, ``Signal processing aspects of computer music--a survey,'' Computer Music Journal, vol. 1, no. 1, pp. 4-37, 1977.

187

J. A. Moorer, ``The use of the phase vocoder in computer music applications,'' Journal of the Audio Engineering Society, vol. 26, pp. 42-45, Jan./Feb. 1978.

188

J. A. Moorer and M. Berger, ``Linear-phase bandsplitting: Theory and applications,'' Journal of the Audio Engineering Society, vol. 34, pp. 143-152, Mar. 1986.

189

M. Mueller, D. P. W. Ellis, A. Klapuri, and G. Richard, ``Signal processing for music analysis,'' IEEE Journal of Selected Topics in Signal Processing, vol. 5, no. 6, pp. 1088-1110, 2011.

190

W. A. Munson and H. C. Montgomery, ``A speech analyzer and synthesizer,'' Journal of the Acoustical Society of America, vol. 22, no. 5, pp. 678-678, 1950.

191

S. H. Nawab and T. F. Quatieri, ``Short-time Fourier transform,'' in Adanced topics in signal processing (J. S. Lim and A. V. Oppenheim, eds.), ch. 6, Prentice-Hall, 1988.

192

S. Nawab, T. Quatieri, and J. Lim, ``Signal reconstruction from short-time Fourier transform magnitude,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-31, pp. 986-998, Aug 1983.

193

A. Ng and A. Horner, ``Iterative combinatorial basis spectra in wavetable matching,'' Journal of the Audio Engineering Society, vol. 50, no. 12, pp. 1054-1063, 2002.

194

Nguyen, ``Near-perfect-reconstruction pseudo-qmf banks,'' IEEE Transactions on Signal Processing, vol. 43, pp. 65-76, Jan. 1994.

195

P. Noll, ``Digital audio coding for visual communications,'' Proc. IEEE, vol. 83, pp. 925-943, June 1995.

196

A. H. Nuttall, ``Some windows with very good sidelobe behavior,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-29, pp. 84-91, Feb 1981.

197

J. C. O'Neill, P. Flandrin, and W. C. Karl, ``Sparse representations with chirplets via maximum likelihood estimation.''
citeseer.nj.nec.com/389174.html.

198

A. V. Oppenheim and R. W. Schafer, Digital Signal Processing,
Englewood Cliffs, NJ: Prentice-Hall, 1975.

199

D. O'Shaughnessy, Speech Communication,
Reading MA: Addison-Wesley, 1987.

200

T. Painter and A. Spanias, ``Perceptual coding of digital audio,'' Proceedings of the IEEE, vol. 88, pp. 451-513, Apr. 2000.

201

A. Papoulis, Probability, Random Variables, and Stochastic Processes,
New York: McGraw-Hill, 1965.

202

A. Papoulis, Signal Analysis,
New York: McGraw-Hill, 1977.

203

A. Papoulis, Probability, Random Variables, and Stochastic Processes,
New York: McGraw-Hill, 1984.

204

T. W. Parks and C. S. Burrus, Digital Filter Design,
New York: John Wiley and Sons, Inc., June 1987,
contains FORTRAN software listings.

205

R. D. Patterson, ``Auditory filter shapes derived with noise stimuli,'' Journal of the Acoustical Society of America, vol. 76, pp. 640-654, Mar. 1982.

206

R. D. Patterson and G. B. Henning, ``Stimulus variability and auditory filter shape,'' Journal of the Acoustical Society of America, vol. 62, pp. 649-663, Sept. 1977.

207

R. D. Patterson, I. Nimmo-Smith, D. L. Weber, and R. Milroy, ``The deterioration of hearing with age: Frequency selectivity, the critical ratio, the audiogram, and speech threshold,'' Journal of the Acoustical Society of America, vol. 72, pp. 1788-1803, Dec. 1982.

208

R. D. Patterson, M. Allerhand, and C. Giguere, ``Time-domain modelling of peripheral auditory processing: A modular architecture and software platform,'' Journal of the Acoustical Society of America, vol. 98, pp. 1890-1894, 1995.

209

V. Peltonen, J. Tuomi, A. Klapuri, J. Huopaniemi, and T. Sorsa, ``Computational auditory scene recognition,'' Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, 2002,
http://www.cs.tut.fi/~klap/iiro/index.html.

210

E. Peterson and F. S. Cooper, ``Peak picker: A bandwidth compression device,'' Journal of the Acoustical Society of America, vol. 29, p. 777, June 1957,
See also the section entitled ``Peak Picking'' in [245, pp. 357].

211

R. Plomp, Aspects of Tone Sensation,
New York: Academic Press, 1976.

212

M. R. Portnoff, ``Implementation of the digital phase vocoder using the fast Fourier transform,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-24, pp. 243-248, June 1976.

213

M. R. Portnoff, ``Time-frequency representation of digital signals and systems based on short-time Fourier analysis,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-28, pp. 55-69, Feb 1980.

214

J. P. Princen and A. B. Bradley, ``Analysis/synthesis filter bank design based on time domain aliasing cancellation,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. 34, pp. 1153-1161, Oct. 1986.

215

M. Puckette, ``Phase-locked vocoder,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), pp. Session 9a, Paper 3, IEEE Press, Oct. 18, 1995.

216

M. Puckette, Theory and Techniques of Electronic Music,
http://crca.ucsd.edu/~msp/techniques/latest/book-html/book.html, 2003.

217

H. Purnhagen and N. Meine, ``Hiln--the MPEG-4 parametric audio coding tools,'' in IEEE Proc. ISCAS 2000, Geneva, vol. 3, pp. 201-204, May 2000.

218

W. Putnam and J. O. Smith, ``Design of fractional delay filters using convex optimization,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 1997,
https://ccrma.stanford.edu/~jos/resample/optfir.pdf.

219

T. Quatieri, S. H. Nawab, and J. S. Lim, ``Frequency sampling of the short-time Fourier-transform magnitude for signal reconstruction,'' Journal of the Optical Society of America, vol. 73, pp. 1523-1526, Nov 1983.

220

T. F. Quatieri and T. E. Hanna, ``Time-scale modification with inconsistent constraints,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), pp. Session 10, Paper 2, IEEE Press, Oct. 18, 1995.

221

T. F. Quatieri and R. J. McAulay, ``Magnitude-only reconstruction using a sinusoidal speech model,'' Proc. IEEE ICASSP-84, vol. 2, pp. 27.6.1-27.6.4, 1984.

222

T. F. Quatieri and R. J. McAulay, ``Speech transformations based on a sinusoidal representation,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-34, pp. 1449-1464, Dec 1986.

223

T. F. Quatieri and R. J. McAulay, ``Audio signal processing based on sinusoidal analysis/synthesis,'' in Applications of DSP to Audio & Acoustics (M. Kahrs and K. Brandenburg, eds.), pp. 343-416, Boston/Dordrecht/London: Kluwer Academic Publishers, 1998.

224

L. R. Rabiner and B. Gold, Theory and Application of Digital Signal Processing,
Prentice-Hall, 1975.

225

L. R. Rabiner and C. M. Rader, eds., Digital Signal Processing,
New York: IEEE Press, 1972.

226

L. R. Rabiner and R. W. Schafer, Digital Processing of Speech Signals,
Prentice-Hall, 1978.

227

L. Rabiner, Fundamentals of Speech Recognition,
Prentice Hall, Apr. 1993.

228

A. Reilly, G. Frazer, and B. Boashash, ``Analytic signal generation--tips and traps,'' IEEE Transactions on Signal Processing, vol. 42, Nov 1994.

229

J. R. Rice, The Approximation of Functions, Volume I,
Reading MA: Addison-Wesley, 1964.

230

D. C. Rife and R. R. Boorstyn, ``Single-tone parameter estimation from discrete-time observations,'' IEEE Transactions on Information Theory, vol. IT-20, pp. 591-598, Sep. 1974.

231

D. C. Rife and R. R. Boorstyn, ``Multiple tone parameter estimation from discrete-time observations,'' Bell System Technical Journal, vol. 55, pp. 1389-1410, Nov. 1976.

232

J.-C. Risset, ``Computer music experiments, 1964--$\ldots$ ,'' Computer Music Journal, vol. 9, no. 1, pp. 11-18, 1985,
Reprinted in [234].

233

J.-C. Risset and M. V. Mathews, ``Analysis of musical-instrument tones,'' Physics Today, vol. 22, no. 2, pp. 23-30, 1969.

234

C. Roads, ed., The Music Machine,
Cambridge, MA: MIT Press, 1989.

235

C. Roads, The Computer Music Tutorial,
Cambridge, MA: MIT Press, 1996.

236

C. Roads and J. Strawn, eds., Foundations of Computer Music,
Cambridge, MA: MIT Press, 1985.

237

C. Roads, S. T. Pope, A. Piccialli, and G. De Poli, eds., Musical Signal Processing,
Netherlands: Swets and Zietlinger, 1997.

238

A. Röbel, ``A new approach to transient processing in the phase vocoder,'' in Proceedings of the Conference on Digital Audio Effects (DAFx-03), Queen Mary, University of London, 2003.

239

X. Rodet and P. Depalle, ``Spectral envelopes and inverse FFT synthesis,'' Proc. 93rd Convention of the Audio Engineering Society, San Francisco, 1992,
Preprint 3393 (H-3).

240

S. Roucos and A. M. Wilgus, ``High quality time scale modification for speech,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Tampa, Florida, pp. 493-496, Mar. 1985,
Introduces the SOLA (Synchronous OverLap-Add) technique for time scale modification.

241

J. Salsman, ``Matlab programming/phase vocoder and encoder,'' 2007,
http://en.wikibooks.org/wiki/MATLAB_Programming/Phase_Vocoder_and_Encoder.

242

L. Savioja, V. Välimäki, and J. O. Smith, ``Audio signal processing using graphics processing units,'' Journal of the Audio Engineering Society, vol. 59, pp. 3-19, Jan. 2011.

243

R. W. Schafer and J. D. Markel, eds., Speech Analysis,
New York: IEEE Press, 1979.

244

C. Schörkhuber and A. Klapuri, ``Constant-Q transform toolbox for music processing,'' in Proc. 7th Sound and Music Conf (SMC-2010), 2010,
http://www.elec.qmul.ac.uk/people/anssik/cqt/smc2010.pdf.

245

M. R. Schroeder, ``Vocoders: Analysis and synthesis of speech (a review of 30 years of applied speech research),'' Proceedings of the IEEE, vol. 54, pp. 720-734, May 1966,
Reprinted in [243, pp. 352-366].

246

X. Serra, A System For Sound Analysis Transformation Synthesis Based on a Deterministic Plus Stochastic Decomposition,
PhD thesis, Stanford University, 1989.

247

X. Serra, ``A computer model for bar percussion instruments,'' in Proceedings of the 1986 International Computer Music Conference, The Hague, pp. 257-262, Computer Music Association, searchable at http://quod.lib.umich.edu/i/icmc/, 1986.

248

X. Serra, ``Musical sound modeling with sinusoids plus noise,'' in Musical Signal Processing (C. Roads, S. Pope, A. Picialli, and G. De Poli, eds.), pp. 91-122, The Netherlands: Swets & Zeitlinger Publishers, June 1997.

249

X. Serra and J. O. Smith, ``Spectral modeling synthesis: A sound analysis/synthesis system based on a deterministic plus stochastic decomposition,'' Computer Music Journal, vol. 14, no. 4, pp. 12-24, 1990,
Software and recent papers URL: //www.iua.upf.es/~xserra/.

250

X. Serra and J. O. Smith, ``Sound-sheet examples for a sound analysis/synthesis system based on a deterministic plus stochastic decomposition,'' Computer Music Journal, vol. 15, pp. 86-87, Spring 1991.

251

M. J. Shailer, B. C. J. Moore, B. R. Glasberg, and N. Watson, ``Auditory filter shapes at 8 and 10 kHz,'' Journal of the Acoustical Society of America, vol. 88, pp. 141-148, July 1990.

252

L. L. Sharf, Statistical Signal Processing, Detection, Estimation, and Time Series Analysis,
Reading MA: Addison-Wesley, 1991.

253

G. Shi, X. Xie, X. Chen, and W. Zhong, ``Recent advances and new design method in nonuniform filter banks,'' in Proceedings of the IEEE International Conference on Comm., Circ. & Sys., vol. 1, pp. 211-215, June 2006.

254

D. J. Shpak and A. Antoniou, ``A generalized Remez method for the design of FIR digital filters,'' IEEE Transactions on Circuits and Systems, vol. CAS-37, pp. 161-174, Feb. 1990,
basis of firgr in the Matlab Filter Design Toolbox.

255

M. Slaney, ``An efficient implementation of the Patterson-Holdsworth auditory filter bank,'' Tech. Rep. 35, Apple Computer, Inc., 1993,
http://www.slaney.org/malcolm/apple/tr35/PattersonsEar.pdf.

256

D. Slepian and H. Pollak, ``Prolate-spheroidal wave functions, Fourier analysis and uncertainty--i,'' Bell System Technical Journal, vol. 40, pp. 34-64, Jan. 1961.

257

A. Smirnov, ``Proto musique concrète--Russian futurism in the 10s and 20s, and early ideas of sonic art and the art of noises,'' in Inventionen 98: 50 Jahre Musique Concrète, Jagerstrasse 23, D-10117 Berlin: DAAD, 1998.

258

J. O. Smith, Techniques for Digital Filter Design and System Identification with Application to the Violin,
PhD thesis, Elec. Engineering Dept., Stanford University (CCRMA), June 1983,
CCRMA Technical Report STAN-M-14, https://ccrma.stanford.edu/STANM/stanms/stanm14/.

259

J. O. Smith, ``Physical modeling using digital waveguides,'' Computer Music Journal, vol. 16, no. 4, pp. 74-91, 1992,
special issue: Physical Modeling of Musical Instruments, Part I. https://ccrma.stanford.edu/~jos/pmudw/.

260

J. O. Smith, Introduction to Matlab and Octave,
https://ccrma.stanford.edu/~jos/matlab/, 2003.

261

J. O. Smith, Introduction to Digital Filters,
https://ccrma.stanford.edu/~jos/filters05/, September 2005.

262

J. O. Smith, Physical Audio Signal Processing: Digital Waveguide Modeling of Musical Instruments and Audio Effects,
https://ccrma.stanford.edu/~jos/pasp/, Dec. 2005.

263

J. O. Smith, Introduction to Digital Filters with Audio Applications,
https://ccrma.stanford.edu/~jos/filters/, Sept. 2007,
online book.

264

J. O. Smith, Mathematics of the Discrete Fourier Transform (DFT), with Audio Applications, Second Edition,
https://ccrma.stanford.edu/~jos/mdft/, Apr. 2007,
online book.

265

J. O. Smith, ``Audio FFT filter banks,'' in Proceedings of the 12th International Conference on Digital Audio Effects (DAFx-09), Como, Italy, September 1-4, Sept. 2009.

266

J. O. Smith, Physical Audio Signal Processing,
https://ccrma.stanford.edu/~jos/pasp/, Dec. 2010,
online book.

267

J. O. Smith, Physical Audio Signal Processing,
https://ccrma.stanford.edu/~jos/pasp/, Dec. 2010,
W3K Publishing, online book.

268

J. O. Smith and J. S. Abel, ``The Bark bilinear transform,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 1995,
Session 8, paper 6, 4 pages. https://ccrma.stanford.edu/~jos/gz/bbtmh.tgz.

269

J. O. Smith and J. S. Abel, ``Bark and ERB bilinear transforms,'' IEEE Transactions on Speech and Audio Processing, pp. 697-708, Nov. 1999.

270

J. O. Smith and P. Gossett, ``A flexible sampling-rate conversion method,'' in Proc. 1984 Int. Conf. Acoustics, Speech, and Signal Processing (ICASSP-84), San Diego, vol. 2, (New York), pp. 19.4.1-19.4.2, IEEE Press, Mar. 1984,
expanded tutorial and associated free software available at the Digital Audio Resampling Home Page: https://ccrma.stanford.edu/~jos/resample/.

271

J. O. Smith and X. Serra, ``PARSHL: A program for the analysis/synthesis of inharmonic sounds based on a sinusoidal representation,'' in Proceedings of the 1987 International Computer Music Conference, Champaign-Urbana, Computer Music Association, searchable at http://quod.lib.umich.edu/i/icmc/, 1987,
Also available as Stanford Music Department Technical Report STAN-M-43. Expanded version available on-line at https://ccrma.stanford.edu/~jos/parshl/.

272

M. J. T. Smith and T. P. Barnwell, ``A unifying framework for analysis/synthesis systems based on maximally decimated filter banks,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Tampa, Florida, (New York), pp. 521-524, IEEE Press, 1985.

273

A. Spanias, T. Painter, and V. Atti, Audio Signal Processing and Coding,
New York: John Wiley and Sons, Inc., 2007.

274

T. Sporer and K. Brandenburg, ``Constraints of filter banks used for perceptual measurement,'' Journal of the Audio Engineering Society, vol. 43, pp. 107-116, Mar. 1995.

275

K. Steiglitz, A Digital Signal Processing Primer with Applications to Audio and Computer Music,
Reading MA: Addison-Wesley, 1996.

276

S. S. Stevens and H. Davis, Hearing: Its Psychology and Physiology,
American Inst. Physics, for the Acoustical Society of America, 1983,
copy of original 1938 edition, http://asa.aip.org/publications.html.

277

T. G. Stockham, ``High speed convolution and correlation,'' Proceedings of the April 26-28 Joint Computer Conference (AFIPS-66, Spring), pp. 129-133, 1966.

278

H. W. Strube, ``Linear prediction on a warped frequency scale,'' Journal of the Acoustical Society of America, vol. 68, no. 4, pp. 1071-1076, 1980.

279

R. D. Strum and D. E. Kirk, First Principles of Discrete Systems and Digital Signal Processing,
Reading MA: Addison-Wesley, 1988.

280

R. K. C. Tan and A. H. J. Lin, ``A time-scale modification algorithm based on the subband time-domain technique for broad-band signal applications,'' Journal of the Audio Engineering Society, vol. 46, pp. 437-449, May 2000.

281

Tech. staff, ``ADEC subroutine description,'' Tech. Rep. 13201, General Electric Co., Syracuse, NY, June 1977.

282

H. Thornburg, On the Detection and Modeling of Transient Audio Signals with Prior Information,
PhD thesis, Elec. Engineering Dept., Stanford University (CCRMA), September 2005.

283

J. Treichler, ``Notes on the design of optimal fir filters,'' tech. rep., Connexions, Sept. 2009,
http://cnx.org/content/col10553/latest/.

284

J. M. Tribolet and R. E. Crochiere, ``Frequency-domain coding of speech,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-27, pp. 512-530, Oct. 1979.

285

C. Truesdell, ``The rational mechanics of flexible or elastic bodies, 1638­1788,'' in Leonardi Euleri Opera Omnia, Ser. 2, XI part 2, 1960.

286

M. Unser, ``Splines: A perfect fit for signal and image processing,'' IEEE Signal Processing Magazine, vol. 16, pp. 22-38, Nov. 1999.

287

P. P. Vaidyanathan, Multirate Systems and Filter Banks,
Prentice-Hall, 1993.

288

V. Välimäki and T. Tolonen, ``Development and calibration of a guitar synthesizer,'' Journal of the Audio Engineering Society, vol. 46, Sept. 1998,
Proceedings of the 103rd Convention of the Audio Engineering Society, New York, Sept. 1997.

289

(Various), Prime-Factor FFT Algorithm,
http://www.wikipedia.org/wiki/Prime-factor_FFT_algorithm, 2003.

290

T. Verma, A Perceptually Based Audio Signal Model With Application to Scalable Audio Compression,
PhD thesis, Elec. Engineering Dept., Stanford University (CCRMA), 2000.

291

M. Vetterli and J. Kovacevic, Wavelets and Subband Coding,
Englewood Cliffs, NJ: Prentice-Hall, 1995.

292

H. L. F. von Helmholtz, Die Lehre von den Tonempfindungen als Physiologische Grundlage für die Theorie der Musik,
Braunschweig: F. Vieweg und Sohn, 1863.

293

H. L. F. von Helmholtz, On the Sensations of Tone as a Physiological Basis for the Theory of Music,
New York: Dover, 1954,
English translation of 1863 (German) edition.

294

R. F. Voss and J. Clarke, ```1/f noise' in music: Music from 1/f noise,'' Journal of the Acoustical Society of America, vol. 63, pp. 258-263, Jan. 1978.

295

A. Wang, ``Instantaneous and frequency-warped techniques for source separation and signal parametrization,'' in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, (New York), IEEE Press, Oct. 1995.

296

A. Wang, ``private communication,'' 2018.

297

P. D. Welch, ``The use of fast Fourier transforms for the estimation of power spectra: A method based on time averaging over short modified periodograms,'' IEEE Transactions on Audio and Electroacoustics, vol. 15, pp. 70-73, 1967,
reprinted in [37] and [225].

298

J. D. Wise, J. R. Caprio, and T. W. Parks, ``Maximum likelihood pitch estimation,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-24, pp. 418-423, Oct. 1976.

299

B. A. Wright, ``Auditory filter asymmetry at 2000 Hz in 80 normal-hearing ears,'' Journal of the Acoustical Society of America, vol. 100, pp. 1717-1721, Sept. 1996.

300

M. Wright and J. O. Smith, ``Open-source matlab tools for interpolation of SDIF sinusoidal synthesis,'' in Proceedings of the 2005 International Computer Music Conference, Sept., Barcelona, Spain, pp. 632-635, Sept. 5-9 2005.

301

D. T. Yeh, Digital Implementation of Musical Distortion Circuits by Analysis and Simulation,
PhD thesis, CCRMA, Elec. Engineering Dept., Stanford University (CCRMA), June 2009,
https://ccrma.stanford.edu/~dtyeh/.

302

M. M. J. Yekta, ``Equivalence of the Lagrange interpolator for uniformly sampled signals and the scaled binomially windowed shifted sinc function,'' Digital Signal Processing, vol. 19, pp. 838-842, Sept. 2009.

303

Z. Zijing and Y. Yun, ``A simple design method for nonuniform cosine modulated filter banks,'' in IEEE International Symposium on Microwave, Antenna, Prop., & EMC Tech. for Wireless Comm., pp. 1052-1055, 2007.

304

U. Zölzer, ed., DAFX--Digital Audio Effects,
West Sussex, England: John Wiley and Sons, LTD, 2002,
http://www.dafx.de/.

305

E. Zwicker and H. Fastl, Psychoacoustics, Facts and Models,
Berlin: Springer Verlag, 1990,
see also later 1999 edition.

306

E. Zwicker and H. Fastl, Psychoacoustics: Facts and Models,
Berlin: Springer Verlag, 1999,
second updated edition, 80pp., CD-ROM/softcover.

307

E. Zwicker and B. Scharf, ``A model of loudness summation,'' Psych. Rev., vol. 72, pp. 3-26, 1965.

308

E. Zwicker and E. Terhardt, ``Analytical expressions for critical band rate and critical bandwidth as a function of frequency,'' Journal of the Acoustical Society of America, vol. 68, pp. 1523-1525, 1980.