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Bibliography

1
M. Abramowitz and I. A. Stegun, eds., Handbook of Mathematical Functions,
New York: Dover, 1965.

2
R. Agarwal and C. S. Burrus, ``Number theoretic transforms to implement fast digital convolution,'' Proceedings of the IEEE, vol. 63, pp. 550-560, Apr. 1975,
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3
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4
L. Beranek, Concert Halls and Opera Houses: Music, Acoustics, and Architecture,
Berlin: Springer-Verlag, 2004.

5
M. Bosi and R. E. Goldberg, Introduction to Digital Audio Coding and Standards,
Boston: Kluwer Academic Publishers, 2003.

6
L. Bosse, ``An experimental high fidelity perceptual audio coder,'' tech. rep., Elec. Engineering Dept., Stanford University (CCRMA), Mar. 1998,
Music 420 Project Report, http://ccrma.stanford.edu/~jos/bosse/.

7
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.

8
R. Bristow-Johnson, ``Tutorial on floating-point versus fixed-point,'' Audio Engineering Society Convention, Oct. 2008.

9
C. S. Burrus, ``Notes on the FFT,'' Mar. 1990.

10
C. S. Burrus and T. W. Parks, DFT/FFT and Convolution Algorithms,
New York: John Wiley and Sons, Inc., 1985.

11
J. P. Campbell Jr., T. E. Tremain, and V. C. Welch, ``The proposed federal standard 1016 4800 bps voice coder: CELP,'' Speech Technology Magazine, pp. 58-64, April-May 1990.

12
D. C. Champeney, A Handbook of Fourier Theorems,
Cambridge University Press, 1987.

13
D. G. Childers, ed., Modern Spectrum Analysis,
New York: IEEE Press, 1978.

14
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 [62].

15
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16
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17
J. Dattorro, ``The implementation of recursive digital filters for high-fidelity audio,'' Journal of the Audio Engineering Society, vol. 36, pp. 851-878, Nov. 1988,
Comments, ibid. (Letters to the Editor), vol. 37, p. 486 (1989 June); Comments, ibid. (Letters to the Editor), vol. 38, pp. 149-151 (1990 Mar.).

18
J. R. Deller Jr., J. G. Proakis, and J. H. Hansen, Discrete-Time Processing of Speech Signals,
New York: Macmillan, 1993.

19
P. A. M. Dirac, The Principles of Quantum Mechanics, Fourth Edition,
New York: Oxford University Press, 1958-2001.

20
DSP Committee, ed., Programs for Digital Signal Processing,
New York: IEEE Press, 1979.

21
P. Duhamel and M. Vetterli, ``Improved Fourier and Hartley algorithms with application to cyclic convolution of real data,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. 35, pp. 818-824, June 1987.

22
P. Duhamel and M. Vetterli, ``Fast Fourier transforms: A tutorial review and state of the art,'' Signal Processing, vol. 19, pp. 259-299, Apr. 1990.

23
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/.

24
B. Gold and C. M. Rader, Digital Processing of Signals,
New York: McGraw-Hill, 1969.

25
G. H. Golub and C. F. Van Loan, Matrix Computations, 2nd Edition,
Baltimore: The Johns Hopkins University Press, 1989.

26
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addendum: ibid. 22 (2), 373-375 (1960).

27
R. M. Gray and L. D. Davisson, An Introduction to Statistical Signal Processing,
Cambridge University Press, 2003,
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28
J. Gullberg, Mathematics From the Birth of Numbers,
New York: Norton and Co., 1997,
[Qa21.G78 1996] ISBN 0-393-04002-X.

29
W. Hartmann, Signals, Sound, and Sensation,
New York: AIP Press, 1997,
647 pp., 221 illustrations, hardcover.

30
M. Heideman, D. Johnson, and C. S. Burrus, ``Gauss and the history of the FFT,'' IEEE Signal Processing Magazine, vol. 1, pp. 14-21, Oct. 1984,
also in the Archive for History of Exact Sciences, vol. 34, no. 3, pp. 265-277, 1985.

31
S. G. Johnson and M. Frigo, ``A modified split-radix FFT with fewer arithmetic operations,'' IEEE Transactions on Signal Processing, vol. 55, pp. 111-119, Jan. 2007.

32
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Englewood Cliffs, NJ: Prentice-Hall, Apr. 2000.

33
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Englewood Cliffs, NJ: Prentice-Hall, 1993.

34
S. M. Kay, Fundamentals of Statistical Signal Processing, Volume II: Detection Theory,
Englewood Cliffs, NJ: Prentice-Hall, 1998.

35
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also in [43].

36
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.

37
W. R. LePage, Complex Variables and the Laplace Transform for Engineers,
New York: Dover, 1961.

38
M. J. Lighthill, Introduction to Fourier Analysis,
Cambridge University Press, Jan. 1958.

39
L. Ljung and T. L. Soderstrom, Theory and Practice of Recursive Identification,
Cambridge, MA: MIT Press, 1983.

40
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41
J. D. Markel and A. H. Gray, Linear Prediction of Speech,
New York: Springer Verlag, 1976.

42
M. V. Mathews, The Technology of Computer Music,
Cambridge, MA: MIT Press, 1969.

43
J. H. McClellan and C. M. Rader, Number Theory in Digital Signal Processing,
Englewood Cliffs, NJ: Prentice-Hall, 1979.

44
J. H. McClellan, R. W. Schafer, and M. A. Yoder, DSP First: A Multimedia Approach,
Englewood Cliffs, NJ: Prentice-Hall, 1998,
Tk5102.M388.

45
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New York: Academic Press, 1997.

46
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47
B. Noble, Applied Linear Algebra,
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48
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49
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Reading MA: Addison-Wesley, 1987.

50
T. Painter and A. Spanias, ``Perceptual coding of digital audio,'' Proceedings of the IEEE, vol. 88, pp. 451-513, Apr. 2000.

51
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52
A. Papoulis, Signal Analysis,
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53
T. W. Parks and C. S. Burrus, Digital Filter Design,
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contains FORTRAN software listings.

54
A. D. Pierce, Acoustics,
American Institute of Physics, for the Acoustical Society of America, 1989,
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55
J. R. Pierce, ``private communication,'' 1991.

56
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57
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New York: IEEE Press, 1972.

58
L. R. Rabiner and R. W. Schafer, Digital Processing of Speech Signals,
Prentice-Hall, 1978.

59
L. R. Rabiner, R. W. Schafer, and C. M. Rader, ``The chirp z-transform algorithm and its application,'' Bell System Technical Journal, vol. 48, pp. 1249-1292, 1969,
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60
C. Roads, ed., The Music Machine,
Cambridge, MA: MIT Press, 1989.

61
C. Roads, The Computer Music Tutorial,
Cambridge, MA: MIT Press, 1996.

62
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63
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64
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65
L. L. Sharf, Statistical Signal Processing, Detection, Estimation, and Time Series Analysis,
Reading MA: Addison-Wesley, 1991.

66
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,
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67
J. O. Smith, Introduction to Matlab and Octave,
http://ccrma.stanford.edu/~jos/matlab/, 2003.

68
J. O. Smith, Introduction to Digital Filters with Audio Applications,
http://ccrma.stanford.edu/~jos/filters/, Sept. 2007,
online book.

69
J. O. Smith, Physical Audio Signal Processing,
http://ccrma.stanford.edu/~jos/pasp/, 2010,
online book.

70
J. O. Smith, Spectral Audio Signal Processing,
http://ccrma.stanford.edu/~jos/sasp/, Mar. 2010,
online book.

71
J. O. Smith and J. S. Abel, ``Bark and ERB bilinear transforms,'' IEEE Transactions on Speech and Audio Processing, pp. 697-708, Nov. 1999.

72
J. O. Smith and P. Gossett, ``A flexible sampling-rate conversion method,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, 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: http://ccrma.stanford.edu/~jos/resample/.

73
H. V. Sorenson, M. T. Heideman, and C. S. Burrus, ``On calculating the split-radix FFT,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-34, pp. 152-156, Feb. 1986.

74
H. V. Sorenson, D. L. Jones, M. T. Heideman, and C. S. Burrus, ``Real-valued fast Fourier transform algorithms,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-35, pp. 849-863, June 1987.

75
K. Steiglitz, A Digital Signal Processing Primer with Applications to Audio and Computer Music,
Reading MA: Addison-Wesley, 1996.

76
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77
T. Stilson, Efficiently Variable Algorithms in Virtual-Analog Music Synthesis--A Root-Locus Perspective,
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78
T. Stilson and J. O. Smith, ``Alias-free synthesis of classic analog waveforms,'' in Proceedings of the 1996 International Computer Music Conference, Hong Kong, pp. 332-335, Computer Music Association, 1996,
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79
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80
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81
M. Unser, ``Splines: A perfect fit for signal and image processing,'' IEEE Signal Processing Magazine, vol. 16, pp. 22-38, Nov. 1999.

82
V. Välimäki, Discrete-Time Modeling of Acoustic Tubes Using Fractional Delay Filters,
PhD thesis, Report no. 37, Helsinki University of Technology, Faculty of Electrical Engineering, Laboratory of Acoustic and Audio Signal Processing, Espoo, Finland, Dec. 1995,
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83
(Various), Prime-Factor FFT Algorithm,
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84
G. N. Watson, A Treatise on the Theory of Bessel Functions,
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85
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 [13] and [57].

86
U. Zölzer, ed., DAFX--Digital Audio Effects,
West Sussex, England: John Wiley and Sons, LTD, 2002,
http://www.dafx.de/.

87
E. Zwicker and H. Fastl, Psychoacoustics: Facts and Models,
Berlin: Springer Verlag, 1999,
second updated edition, 80pp., CD-ROM/softcover.



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