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Bibliography

1
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2
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3
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4
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5
R. Bristow-Johnson, ``The equivalence of various methods of computing biquad coefficients for audio parametric equalizers,'' Audio Engineering Society Convention, 1994,
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6
R. Bristow-Johnson, ``Audio EQ cookbook,'' 1998,
http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt.

7
R. Bristow-Johnson, ``DSP trick: Fixed-point dc blocking filter with noise-shaping,'' The Unofficial Comp.DSP Home Page, June 22, 2000,
http://www.dspguru.com/comp.dsp/tricks/alg/dc_block.htm.

8
J. R. Buck, M. Daniel, and A. C. Singer, Computer Explorations in Signals and Systems using Matlab,
Englewood Cliffs, NJ: Prentice-Hall, 1997,
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9
C. S. Burrus and T. W. Parks, ``Time domain design of recursive digital filters,'' IEEE Transactions on Audio and Electroacoustics, vol. 18, pp. 137-141, June 1970.

10
C. S. Burrus, J. H. McClellan, A. V. Oppenheim, T. W. Parks, R. W. Schafer, and H. W. Schuessler, Computer-Based Exercises for Signal Processing Using Matlab,
Englewood Cliffs, NJ: Prentice-Hall, 1994.

11
J. A. Cadzow, ``High performance spectral estimation--a new ARMA method,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-28, pp. 524-529, Oct. 1980.

12
G. C. Carter, ed., Coherence and Time Delay Estimation,
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13
D. C. Champeney, A Handbook of Fourier Theorems,
Cambridge University Press, 1987.

14
R. V. Churchill, Complex Variables and Applications,
New York: McGraw-Hill, 1960.

15
P. Cook and G. Scavone, Synthesis ToolKit in C++, Version 4,
http://ccrma.stanford.edu/ccrma/Software/STK/, 2007,
see also http://ccrma.stanford.edu/~jos/stkintro/.

16
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/.

17
P. R. Cook, ``Non-linear periodic prediction for on-line identification of oscillator characteristics in woodwind instruments,'' in Proceedings of the 1991 International Computer Music Conference, Montreal, pp. 157-160, Computer Music Association, 1991.

18
G. Dahlquist and Å. Björck, Numerical Methods,
Englewood Cliffs, NJ: Prentice-Hall, 1974.

19
J. R. Deller Jr., J. G. Proakis, and J. H. Hansen, Discrete-Time Processing of Speech Signals,
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20
C. A. Desoer and E. S. Kuh, Basic Circuit Theory,
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21
M. Dolson, ``The phase vocoder: A tutorial,'' Computer Music Journal, vol. 10, no. 4, pp. 14-27, 1986.

22
DSP Committee, ed., Programs for Digital Signal Processing,
New York: IEEE Press, 1979.

23
G. Fant, Acoustic Theory of Speech Production,
The Hague: Mouton, 1960.

24
A. Farina, ``Simultaneous measurement of impulse response and distortion with a swept-sine technique,'' 108th Audio Engineering Society Convention, Feb. 19-22, 2000,
Preprint 5093.

25
A. Fettweis, ``Wave digital filters: Theory and practice,'' Proceedings of the IEEE, vol. 74, pp. 270-327, Feb. 1986.

26
J. L. Flanagan and R. M. Golden, ``Phase vocoder,'' Bell System Technical Journal, vol. 45, pp. 1493-1509, Nov. 1966,
Reprinted in [74, pp. 388-404].

27
J. L. Flanagan and L. R. Rabiner, eds., Speech Synthesis,
Stroudsburg, Penn.: Dowden, Hutchinson, and Ross, Inc., 1973.

28
G. F. Franklin, J. D. Powell, and M. L. Workman, Digital Control of Dynamic Systems, Third Edition,
Englewood Cliffs, NJ: Prentice-Hall, 1998.

29
G. H. Golub and C. F. Van Loan, Matrix Computations, 2nd Edition,
Baltimore: The Johns Hopkins University Press, 1989.

30
G. C. Goodwin and R. L. Payne, Dynamic System Identification,
New York: Academic Press, 1977.

31
A. Gräf, ``Interfacing Pure Data with Faust,'' in Proceedings of the 5th International Linux Audio Conference (LAC2007), http://www.kgw.tu-berlin.de/~lac2007/proceedings.shtml, 2007,
http: //www.kgw.tu-berlin.de/~lac2007/papers/lac07_graef.pdf.

32
A. H. Gray and J. D. Markel, ``A normalized digital filter structure,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. ASSP-23, pp. 268-277, June 1975.

33
D. Halliday, R. Resnick, and J. Walker, Extended, Fundamentals of Physics, 6th Edition,
New York: John Wiley and Sons, Inc., 2000.

34
W. J. Hess, Algorithms and Devices for Pitch Determination of Speech-Signals,
Berlin: Springer-Verlag, 1983.

35
H. Järveläinen, V. Välimäki, and M. Karjalainen, ``Audibility of inharmonicity in string instrument sounds, and implications to digital sound synthesis,'' in Proceedings of the 1999 International Computer Music Conference, Beijing, pp. 359-362, Oct. 22-27, 1999,
http://www.acoustics.hut.fi/~hjarvela/publications/.

36
T. Kailath, Lectures on Linear Least-Squares Estimation,
New York: Springer Verlag, 1976.

37
T. Kailath, Linear Systems,
Englewood Cliffs, NJ: Prentice-Hall, 1980.

38
T. Kailath, A. H. Sayed, and B. Hassibi, Linear Estimation,
Englewood Cliffs, NJ: Prentice-Hall, Apr. 2000.

39
J. B. Keller, ``Bowing of violin strings,'' Comm. Pure Applied Math., vol. 6, pp. 483-495, 1953.

40
D. Klatt, ``Software for a cascade/parallel formant synthesizer,'' Journal of the Acoustical Society of America, vol. 67, pp. 13-33, 1980.

41
M. Lang, ``Allpass filter design and applications,'' IEEE Transactions on Signal Processing, vol. 46, no. 9, pp. 2505-2514, 1998.

42
M. Lang and T. I. Laakso, ``Simple and robust method for the design of allpass filters using least-squares phase error criterion,'' IEEE Transactions on Circuits and Systems--I: Fundamental Theory and Applications, vol. 41, no. 1, pp. 40-48, 1994.

43
W. R. LePage, Complex Variables and the Laplace Transform for Engineers,
New York: Dover, 1961.

44
M. J. Lighthill, Introduction to Fourier Analysis,
Cambridge University Press, Jan. 1958.

45
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.

46
L. Ljung and T. L. Soderstrom, Theory and Practice of Recursive Identification,
Cambridge, MA: MIT Press, 1983.

47
J. Makhoul, ``Linear prediction: A tutorial review,'' Proceedings of the IEEE, vol. 63, pp. 561-580, Apr. 1975.

48
J. D. Markel and A. H. Gray, Linear Prediction of Speech,
New York: Springer Verlag, 1976.

49
S. J. Mason, ``Feedback theory--some properties of signal flow graphs,'' Proceedings of the IRE, vol. 41, pp. 1144-1156, Sept. 1953.

50
S. J. Mason, ``Feedback theory--further properties of signal flow graphs,'' Proceedings of the IRE, vol. 44, pp. 920-926, July 1956.

51
M. Mathews and J. O. Smith, ``Methods for synthesizing very high Q parametrically well behaved two pole filters,'' in Proceedings of the Stockholm Musical Acoustics Conference (SMAC-03), http://www.speech.kth.se/smac03/, (Stockholm), Royal Swedish Academy of Music, Aug. 2003,
available online, with sound examples, at http://ccrma.stanford.edu/~jos/smac03maxjos/.

52
J. H. McClellan, R. W. Schafer, and M. A. Yoder, DSP First: A Multimedia Approach,
Englewood Cliffs, NJ: Prentice-Hall, 1998,
Tk5102.M388.

53
F. R. Moore, ``An introduction to the mathematics of digital signal processing, parts I-II,'' Computer Music Journal, vol. 2, no. 1,2, pp. 38-47,38-60, 1978,
available at CCRMA (see http://ccrma.stanford.edu/overview/publications.html).

54
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.

55
P. M. Morse, Vibration and Sound,
http://asa.aip.org/publications.html: American Institute of Physics, for the Acoustical Society of America, 1948,
1st edition 1936, last author's edition 1948, ASA edition 1981.

56
A. W. Nayor and G. R. Sell, Linear Operator Theory in Engineering and Science,
New York: Springer Verlag, 1982.

57
Z. Nehari, Conformal Mapping,
New York: Dover, 1952.

58
B. Noble, Applied Linear Algebra,
Englewood Cliffs, NJ: Prentice-Hall, 1969.

59
A. V. Oppenheim, Discrete-Time Signal Processing,
Englewood Cliffs, NJ: Prentice-Hall, 1989.

60
A. V. Oppenheim and R. W. Schafer, Digital Signal Processing,
Englewood Cliffs, NJ: Prentice-Hall, 1975.

61
Y. Orlarey, A. Gräf, and S. Kersten, ``DSP programming with Faust, Q and SuperCollider,'' in Proceedings of the 4th International Linux Audio Conference (LAC2006), http://lac.zkm.de/2006/proceedings.shtml, 2006,
http://www.grame.fr/pub/lac06.pdf.

62
J. M. Ortega, Numerical Analysis,
New York: Academic Press, 1972.

63
A. Papoulis, Signal Analysis,
New York: McGraw-Hill, 1977.

64
T. W. Parks and C. S. Burrus, Digital Filter Design,
New York: John Wiley and Sons, Inc., June 1987,
contains FORTRAN software listings.

65
M. Puckette, Pure Data (PD),
http://www.puredata.org, July 2004.

66
M. Puckette, Theory and Techniques of Electronic Music,
http://www.worldscibooks.com/compsci/6277.html: World Scientific Press, May 2007,
http://www-crca.ucsd.edu/~msp/techniques.htm.

67
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,
http://ccrma.stanford.edu/~jos/resample/optfir.pdf.

68
L. R. Rabiner and B. Gold, Theory and Application of Digital Signal Processing,
Prentice-Hall, 1975.

69
L. R. Rabiner and R. W. Schafer, Digital Processing of Speech Signals,
Prentice-Hall, 1978.

70
G. R. Reddy and M. N. S. Swamy, ``Digital all-pass filter design through discrete hilbert transform,'' in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Albuquerque, 1998.

71
C. Roads, ed., The Music Machine,
Cambridge, MA: MIT Press, 1989.

72
D. Rocchesso and F. Scalcon, ``Accurate dispersion simulation for piano strings,'' in Proc. Nordic Acoustical Meeting (NAM'96), (Helsinki, Finland), June 12-14 1996,
9 pages.

73
W. Rudin, Principles of Mathematical Analysis,
New York: McGraw-Hill, 1964.

74
R. W. Schafer and J. D. Markel, eds., Speech Analysis,
New York: IEEE Press, 1979.

75
H. Schmid, ``Circuit transposition using signal-flow graphs,'' in Proceedings of the International Symposium Circuits and Systems (ISCAS-2002), Phoenix, AZ, vol. 2, (New York), pp. 25-28, IEEE Press, May 2002.

76
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 [74, pp. 352-366].

77
L. L. Sharf, Statistical Signal Processing, Detection, Estimation, and Time Series Analysis,
Reading MA: Addison-Wesley, 1991.

78
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, http://ccrma.stanford.edu/STANM/STANM/stanm14/.

79
J. O. Smith, ``Introduction to digital filter theory,'' in Digital Audio Signal Processing: An Anthology (J. Strawn, ed.), Los Altos, California: William Kaufmann, Inc., 1985,
(out of print). Book reprint available from ICMA at http://www.computermusic.org/. Original version available as Stanford CCRMA Tech. Report STAN-M-20, April 1985. A shortened version appears in [71].

80
J. O. Smith, ``Music applications of digital waveguides,'' Tech. Rep. STAN-M-39, CCRMA, Music Department, Stanford University, 1987,
CCRMA Technical Report STAN-M-39, http://ccrma.stanford.edu/STANM/stanm39/.

81
J. O. Smith, ``Principles of digital waveguide models of musical instruments,'' in Applications of Digital Signal Processing to Audio and Acoustics (M. Kahrs and K. Brandenburg, eds.), pp. 417-466, Boston/Dordrecht/London: Kluwer Academic Publishers, 1998.

82
J. O. Smith, Introduction to Matlab and Octave,
http://ccrma.stanford.edu/~jos/matlab/, 2003.

83
J. O. Smith, Introduction to Digital Filters with Audio Applications,
http://ccrma.stanford.edu/~jos/filters/, Sept. 2007,
online book.

84
J. O. Smith, Mathematics of the Discrete Fourier Transform (DFT), with Audio Applications, Second Edition,
http://ccrma.stanford.edu/~jos/mdft/, Apr. 2007,
online book.

85
J. O. Smith, ``Virtual electric guitars and effects using Faust and Octave,'' in Proceedings of the 6th International Linux Audio Conference (LAC2008), http://lac.linuxaudio.org/, 2008,
paper: http://ccrma.stanford.edu/realsimple/faust_strings/faust_strings.pdf, presentation overheads: http://ccrma.stanford.edu/~jos/pdf/LAC2008-jos.pdf, supporting website: http://ccrma.stanford.edu/realsimple/faust_strings/.

86
J. O. Smith, Physical Audio Signal Processing,
http://ccrma.stanford.edu/~jos/pasp/, 2010,
online book.

87
J. O. Smith, Spectral Audio Signal Processing,
http://ccrma.stanford.edu/~jos/sasp/, Mar. 2010,
online book.

88
J. O. Smith and J. S. Abel, ``Bark and ERB bilinear transforms,'' IEEE Transactions on Speech and Audio Processing, pp. 697-708, Nov. 1999.

89
J. O. Smith and J. B. Angell, ``A constant-gain digital resonator tuned by a single coefficient,'' Computer Music Journal, vol. 6, no. 4, pp. 36-40, 1982.

90
J. O. Smith and P. R. Cook, ``The second-order digital waveguide oscillator,'' in Proceedings of the 1992 International Computer Music Conference, San Jose, pp. 150-153, Computer Music Association, 1992,
http://ccrma.stanford.edu/~jos/wgo/.

91
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/.

92
A. S. Spanias, ``Speech coding: A tutorial review,'' Proceedings of the IEEE, vol. 82, Oct. 1994.

93
K. Steiglitz, A Digital Signal Processing Primer with Applications to Audio and Computer Music,
Reading MA: Addison-Wesley, 1996.

94
K. Steiglitz, ``A note on constant-gain digital resonators,'' Computer Music Journal, vol. 18, no. 4, pp. 8-10, 1994.

95
T. Stilson and J. O. Smith, ``Analyzing the Moog VCF with considerations for digital implementation,'' in Proceedings of the 1996 International Computer Music Conference, Hong Kong, Computer Music Association, 1996,
http://ccrma.stanford.edu/~stilti/.

96
J. C. Strikwerda, Finite Difference Schemes and Partial Differential Equations,
Pacific Grove, CA: Wadsworth and Brooks, 1989.

97
T. Tolonen, V. Välimäki, and M. Karjalainen, ``Modeling of tension modulation nonlinearity in plucked strings,'' IEEE Transactions on Speech and Audio Processing, vol. SAP-8, pp. 300-310, May 2000.

98
P. P. Vaidyanathan, Multirate Systems and Filter Banks,
Prentice-Hall, 1993.

99
S. A. Van Duyne and J. O. Smith, ``Implementation of a variable pick-up point on a waveguide string model with FM/AM applications,'' in Proceedings of the 1992 International Computer Music Conference, San Jose, pp. 154-157, Computer Music Association, 1992.

100
B. Yegnanarayana, ``Design of recursive group-delay filters by autoregressive modeling,'' IEEE Transactions on Acoustics, Speech, Signal Processing, vol. 30, pp. 632-637, Aug. 1982.

101
D. Yeh and J. O. Smith III, ``Discretization of the '59 Fender Bassman tone stack,'' Proceedings of the Conference on Digital Audio Effects (DAFx-06), Montreal, Canada, Sept. 2006,
http://www.dafx.de/.

102
L. A. Zadeh and C. A. Desoer, Linear System Theory: The State Space Approach,
New York: McGraw-Hill, 1963,
reprinted by Krieger, 1979.

103
U. Zölzer, Digital Audio Signal Processing,
New York: John Wiley and Sons, Inc., 1999.



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