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This book is the only real reference for filter banks and multirate systems, as opposed to being a tutorial.
Peter Kootsookosnotes: this book is most certainly an excellent book on multi-rate signal processing, but it came out right before perfect reconstruction filter banks hit the streets. Multirate Systems and Filter Banks by P. P. Vaidyanathan covers this issue.
Excellent reference work, but assumes you know a fair amount to begin with. [Phil Lapsley]
This is an updated version of the original, with some old material deleted and lots of new material added.
An introduction to signal processing methods which have many applications including speech analysis, image processing, and oil exploration. The author uses optimum Wiener filtering and least-squares estimation concepts as unifying themes and includes subroutines for FORTRAN and C. [Juergen Kahrs, jkahrs@castor.atlas.de]
Thomas Parsons, Voice and Speech Processing, McGraw-Hill, 1987, ISBN 0-07-048541-0.
The book is also available on-line at http://www.nr.com.
J. G. Proakis and D. G. Manolakis, Digital Signal Processing: Principles, Algorithms, and Applications, MacMillan Publishing, New York, NY, 1992, ISBN 0-02-396815-X.
L. R. Rabiner and R. W. Schafer, Digital Processing of Speech Signals, Prentice Hall, 1978, ISBN 0-13-213603-1.
S. D. Stearns and R. A. David, Signal Processing Algorithms, Prentice Hall, Eaglewood Cliffs, NJ, 1988. ISBN
P. P. Vaidyanathan, Multirate Systems and Filter Banks, Prentice-Hall. 911 pp. ISBN 0-13-605718-7.
R.A. Monzingo and T.W. Miller, Introduction to Adaptive Arrays, John Wiley and Sons, NY, 1980.
Perhaps the classic overview paper for discrete-time windows. It discusses some 15 different classes of windows including their spectral responses and the reasons for their development. [Brian Evans, bevans@ece.utexas.edu]There are several typos in the above paper. The errors are corrected in:
An elegant method for designing a time-discrete solution for realization of a spectral window which is ideal from an energy concentration viewpoint. This window is one that concentrates the maximum amount of energy in a specified bandwidth and hence provides optimal spectral resolution. Unlike the Kaiser window, this window is a discrete-time realization having the same objectives as the continuous-time prolate spheroidal function; at the expense of not having a closed form solution. [Joe Campbell, jpcampb@afterlife.ncsc.mil]
In his classic 1982 paper, David Thompson proposes the powerful multiple-window method, which is an elegant and robust technique for spectrum estimation. Based on the Cramer representation, Thompson's method is nonparametric, consistent, efficient, and optimally suited for finite data samples. In addition, it has excellent bias control and stability, provides an analysis of variance test for line components, and finally, works very well in many practical applications. Unfortunately, his important work has been neglected in many textbooks and graduate courses on statistical signal processing. [Dong Wei, wei@vision.ece.utexas.edu, and Brian Evans, bevans@ece.utexas.edu]
Hal Chamberlin, Musical Applications of Microprocessors, 2nd Ed., Hayden Book Company, 1985.
Recommended. [Juhana Kouhia, jk87377@cc.tut.fi]
Contains article on analysis/synthesis by Strawn, recommended; also an another article maybe by J.A. Moorer [Juhana Kouhia, jk87377@cc.tut.fi]
John Strawn, ed., Digital Audio Engineering, 144 pages, A-R Editions. ISBN 0-86576-087-X.
Contains J.A. Moorer's classic "About This Reverb Business..." and contains an article which gives a code for Phase Vocoder -- great tool for EQ, for Pitchshifter and more [Juhana Kouhia, jk87377@cc.tut.fi]
John Strawn, ed., Digital Audio Signal Processing, 283 pages, ISBN 0-86576-082-9, pub: A-R Editions.
Recommended. [Quinn Jensen, jensenq@qcj.icon.com]
David Cope, "Computer Analysis of Musical Style"
Dexter Morrill and Rick Taube, "A Little Book of Computer Music Instruments"
Ok article, but you have to know basic DSP operations. [Juhana Kouhia, jk87377@cc.tut.fi]
Check more articles from Journal of the Audio Engineering Society (JAES), for example more articles by Strawn.
[The above is largely from Quinn Jensen, jensenq@qcj.icon.com; Juhana Kouhia, jk87377@cc.tut.fi; William Alves, alves@calvin.usc.edu; and Paul A Simoneau, pas1@kepler.unh.edu]
A. Bateman and W. Yates, Digital Signal Processing Design, Computer Science Press, MD, 1989.
R. Chassaing, Digital Signal Processing with C and the TMS320C30, Wiley, NY, 1992.
R. Chassaing and D. W. Horning, Digital Signal Processing with the TMS320C25, Wiley, NY, 1990.
R. Chassaing, DSP Applications Using C and the TMS320C6x DSK, Wiley, NY, ISBN 0471207543, 2002.
J. Datta, B. Karley, J. Lane, and J. Norwood, DSP Filter Cookbook, Prompt, 2000.Updated!
Y. Dote, Servo Motor and Motion Control Using Digital Signal Processors, Prentice Hall, NJ, 1990.
P. Embree, C Algorithms for Real-Time DSP, Prentice Hall, 1995.Updated!
R. Higgins, Digital Signal Processing in VLSI, Prentice Hall, NJ, 1990. ISBN 0-13-212887-X.
This is a comprehensive entry-level tutorial for anybody interested in processing of digital sound. Warning: This reflects my at-the-time knowledge, and is not always 100 % correct. Yehar's Digital Sound Processing Tutorial for the Braindead or http://www.iki.fi./o/dsp
[Brian Evans, bevans@ece.utexas.edu; Andreas Spanias, spanias@asu.edu]
Prof. Brian Evans: Real-time DSP course online at http://www.ece.utexas.edu/~bevans/courses/realtime/.
TechOnLine (http://www.techonline.com/): Courses on various topics.
Engineering Productivity Tools (http://www.engineeringproductivitytools.com/stuff/T0001/): The FFT Demystified. V2.1
BORES Signal Processing DSP course. (http://www.bores.com/courses/intro/index.htm): Introduction courses to DSP.
TI has a centralized training site where DSP designers can access all of TI's training webcasts, workshops and seminars. It can be found at www.dspvillage.ti.com/trainingpr2. It covers TI DSP, tools, software and applications. Analog training is also included.
TI also has a site designed to help new DSP users (primarily new TI DSP users) get started with their designs: http://www.dspvillage.ti.com/cocostu.
There are close to 100 "M-files" that implement various functions. Some of them are quite simple and are based on existing MATLAB M-files. But a great many of them has been created from scratch. We also prepared a lab manual (in TEX format) for the 7 simulations which the students perform as the lab component of this course. The topics of these simulations are:
The complete manual in Postscript format is available at ftp://ftp.mathworks.com/pub/contrib/v4/misc/comm_tbx/comm_tbx.manual.ps. [Mehmet Zeytinoglu, mzeytin@ee.ryerson.ca]
Note: FOR STUDENTS: A student version of Mathematica is available. It includes a copy of the reference manual. The only drawbacks to the student version are that the floating point coprocessor is disabled and that upgrades cannot be ordered.
Version 3.0 of the SPP (an "overhauled version of 2.x" according to the author) is available commercially in two products: the Signals and Systems Pack from Wolfram Research, and a book entitled "Mathematica Notebooks to Accompany Contemporary Linear Systems Using MATLAB" from PWS Publishing company.
The following Mathematica notebooks (from Julius Smith, jos@ccrma.stanford.edu) can be ftped from ccrma-ftp.stanford.edu:
(There are other DSP related items in pub/DSP on ccrma-ftp; see other sections of this FAQ for details).
The kit is located in the at:
A sample kit of sound-bites is available as: ftp://crl.dec.com/pub/DEC/AF/AF2R2-other.tar
Author: Dr. Shalom Halevy, shalevy@mathwizards.com, PO BOX 22564, San Diego, CA 92192 (619) 552-9031 USA (Tel/FAX) http://www.mathwizards.com.
The current version of this software demonstrates continuous time convolution, discrete time, and circular convolution along with cross-correlation.
Dr. Kurt Kosbar
117 Electrical Engineering Building
University of Missouri - Rolla
Rolla, Missouri, USA 65401, phone: (573) 341-4894
e-mail: kk@ee.umr.edu
Organizations without Internet access can obtain Ptolemy, without support, from ILP. This is often a more stable, less featured version than is available by FTP.
EECS/ERL Industrial Liaison Program Office
Software Distribution
205 Cory Hall
University of California, Berkeley
Berkeley, CA 94720
(510) 643-6687
email: ilpsoftware@eecs.berkeley.edu
This includes printed documentation, including installation instructions, a user's guide, and manual pages. A handling fee will be charged.
A public domain version of the same Naval Research Lab text to phoneme rules can be obtained from:
ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/syntheses/english2phoneme.tar.gz
The comp.speech FTP site includes a speech synthesis directory at ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis. The main package is "rsynth" which is a complete text to speech synthesis system. Several component packages are also present. "textnorm" converts non-words such as digit strings into words (e.g. 1000 to ONE THOUSAND). "english2phoneme" does some of the same but its main functionality is to guess an appropriate phoneme sequence for each word. "klatt" takes a parametric form that describes each phoneme and converts it to a waveform. Other packages exist in the same directory to edit and visualise the klatt parameters. [Tony Robinson, ajr@softsound.com]
The source for the programs (meteor.p, form.p, meteor.c, and form.c) and the METEOR paper as a postscript file may be found at http://www. music.Princeton.edu/classes/class.html. The programs were originally written in Pascal and then evidentally run through p2c to produce the C versions; all the necessary Pascal library stuff is included in the C code and they built error-free out of the box for me on an SGI machine.
There is no manual. The paper includes instructions on running the programs. [Steve Clift, clift@mail.anacapa.net]
Weimin Liu has created a Windows 95 interface to the Meteor program, which can be downloaded from http://www.nyx.net/~wliu/filter.html.
The packages are available from http://www.tsp.ece.mcgill.ca/Docs/Software/FilterDesign/FilterDesign.html or directly via anonymous ftp from ftp://ftp.tsp.ece.mcgill.ca/TSP/FilterDesign/.
Another package, libtsp, is a library of C-language routines for signal processing. The package is available from http://www.tsp.ece.mcgill.ca/reports/Software/libtsp/libtsp.html or directly via anonymous ftp from ftp://ftp.tsp.ece.mcgill.ca/pub/libtsp/ [Peter Kabal, kabal@ECE.McGill.CA]
This program was created and tested using Borland C++ 2.0. This requires a pretty reasonable C++ compiler - it is reported that QuickC (not C++) won't do it. [Witold Waldman, from Charles Owen at mgcbo@uxa.ecn.bgu.au; also Andrew Ukrainec, ukrainec@InfoUkes.com]
This program will output to an ASCII file the window coefficients that can be easily dumped to an EPROM or included in a program. It also generates both time and frequency domain graphs so that the user can visually verify the widow record length and spectral content. I will gladly provide any interested parties with my MATLAB code.
Tod M. Schuck
Lockheed Martin NE&SS
Moorestown, NJ 08060
e-mail: tod.m.schuck(no spam)@lmco.com
We have released a set of Matlab packages to optimize the following characteristics of analog filter designs simultaneously:
subject to constraints on the same characteristics. The Matlab packages take about 10 seconds for fourth-order filters and 3 minutes for eighth-order filters to run on a 167-MHz Sun Ultra-2 workstation.
We use the symbolic mathematics environment Mathematica to describe the constrained non-linear optimization problem formally, derive the gradients of the cost function and constraints, and synthesize the Matlab code to perform the optimization. In the public release, we provide the Matlab to optimize analog IIR filters of fourth, sixth, and eighth orders. Using the Mathematica formulation, designers can add new measures and constraints, such as capacitance spread for integrated circuit layout, and regenerate the Matlab code.
We describe the framework in [1]. An earlier version of the framework is described in [2]. We plan to extend this framework to digital IIR filters.
[1] N. Damera-Venkata, B. L. Evans, M. D. Lutovac, and D. V. Tosic, Joint Optimization of Multiple Behavioral and Implementation Properties of Analog Filter Designs, Proc. IEEE Int. Sym. on Circuits and Systems, Monterey, CA, May 31 - Jun. 3, 1998, vol. 6, pp. 286-289. http://www.ece.utexas.edu/~bevans/papers/1998/filter_optimization/.
[2] B. L. Evans, D. R. Firth, K. D. White, and E. A. Lee, Automatic Generation of Programs That Jointly Optimize Characteristics of Analog Filter Designs, Proc. of European Conf. on Circuit Theory and Design, Istanbul, Turkey, August 27-31, 1995, pp. 1047-1050. http://ptolemy.eecs.berkeley.edu/publications/papers/95/filter_design_ecctd95/
[Brian Evans, bevans@combo.ece.utexas.edu]
[Grant Griffin, grant.griffin@iowegian.com]
Alex has made these recipes available here: http://unicorn.us.com/alex/2polefilters.html
The recipes cover Butterworth, Critically-Damped, and Bessel filters. Alex also includes test results; i.e., plots of actual frequency response and step-function temporal response for each filter.
Musicdsp.org is a collection of data gathered for the music dsp community. It includes code for wavetable synthesis, dithering, guitar feedback, and many other effects and algorithms.
[Steve Horne, steve@lurking.demon.co.uk]