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Subject: comp.dsp FAQ [1 of 4]
This article was archived around: NNTP-Posting-Wed, 11 Apr 2007 13:15:48 -0500
Last-modified: Wed Apr 11 2007
FAQs (Frequently asked questions with answers) on Digital Signal Processing
The world-wide web version of the comp.dsp FAQ is maintained and sponsored
by Berkeley Design Technology, Inc. For information on BDTI, visit the
BDTI home page at http://www.bdti.com.
Version date: Apr 11, 2007
- Seth Benton, FAQ maintainer
0. What is comp.dsp?
0.1 Relevant links
0.2 Versions of the comp.dsp FAQ
0.3 DISCLAIMER OF WARRANTY
0.4 Redistribution permission
0.5 Note on the list of manufacturers, addresses, and telephone
1. General DSP
1.1 DSP book and article references
1.1.1 Bibles of DSP theory
1.1.2 Adaptive signal processing
1.1.3 Array signal processing
1.1.4 Windowing articles
1.1.5 Digital audio effects processing
1.1.6 Digital signal processing implementation
1.1.7 Free online books
1.2 DSP training
1.2.1 Courses on DSP
1.2.2 On-Line courses on DSP
1.3 Where can I get free software for general DSP?
1.3.1 DSP packages for MATLAB
1.3.2 DSP packages for Mathematica
1.3.3 Other DSP libraries
1.3.4 DSP software
1.3.5 Text to Speech Conversion Software
1.3.6 Filter design software
1.3.7 Audio effects
2. Algorithms and standards
2.1 Where can I get public domain algorithms for DSP?
2.2 What are CELP and LPC? Where can I get source for them?
2.3 What is ADPCM? Where can I get source for it?
2.4 What is GSM? Where can I get source for it?
2.5 How does pitch perception work, and how do I implement it?
2.6 What standards exist for digital audio? What is AES/EBU? What
2.6.1 Where can I get copies of ITU (formerly CCITT)
2.6.2 What standards are there for digital audio?
2.7 What is mu-law encoding? Where can I get source for it?
2.8 How can I do CD <=> DAT sample rate conversion?
2.9 What are wavelets?
2.9.1 What are wavelets? Where can I get more
2.9.2 What are some good books and papers on
2.9.3 Where can I get some software for wavelets?
2.10 How do I calculate the coefficients for a Hilbert
2.11 Algorithm implementation: floating-point versus fixed-point
3. Programmable DSP chips and their software
3.1 What are the available DSP chips and chip architectures?
3.2 What is the difference between a DSP and a microprocessor?
3.3 Software for Analog Devices DSPs
3.3.1 Where can I get a C compiler for the ADSP-21xx
3.3.2 Where can I get tools for the ADSP-21xxx?
3.3.3 Where can I get an assembler for the ADSP-2105?
3.3.4 Where can I get algorithms or libraries for
Analog Devices DSPs?
3.4 Software for Agere Systems (Formerly Lucent Technologies) DSPs
3.5 Software for Motorola DSPs
3.5.1 Where can I get a free assembler for the
3.5.2 Where can I get a free C compiler for the
3.5.3 Where can I get a disassembler for the Motorola
3.5.4 Where can I get algorithms and libraries for
3.5.5 Where can I get NeXT-compatible Motorola
3.5.6 Where can I get emulators for the 68HC11 (6811)
3.6 Software for Texas Instruments DSPs
3.6.1 Where can I get free algorithms or libraries
for TI DSPs?
3.6.2 Where can I get free development tools for TI
3.6.3 Where can I get a free C compiler for the TI
3.6.4 Where can I get a free assembler for the TI
3.6.5 Where can I get a free simulator for the TI
3.6.6 What is Tick? Where can I get it?
4. DSP development boards
5. Operating Systems
Previous section (Overview) Next section (1)
Q0: What is comp.dsp?
Comp.dsp is a worldwide Usenet news group that is used to discuss
various aspects of digital signal processing. It is unmoderated,
though we try to keep the signal to noise ratio up :-). If you
need to ask a question that isn't in the FAQ, and can't figure out
how to post, consult news.newusers.questions.
Q0.1: Relevant links
Other relevant news groups are:
Relevant FAQs are:
* Higher-order statistics FAQ
* comp.compression FAQ
* comp.realtime FAQ
* comp.speech FAQ
* Audio sampling FAQ
There is an index of DSP-related resources, books, discussion
Other relevant links:
Q0.2: Versions of the comp.dsp FAQ
If you're reading this via the World Wide Web:
Click on http://www.bdti.com/faq/dsp_faq.zip to download a
compressed HTML version of the FAQ.
Click on http://www.bdti.com/faq/dsp_faq.asc.zip to download a
compressed ASCII version of the FAQ.
If you're reading this as ASCII text:
Get with the program and get a web browser. The FAQ is available
on World Wide Web with a much nicer interface. This is especially
true for information presented in tabular form. Try:
Q0.3: DISCLAIMER OF WARRANTY
BERKELEY DESIGN TECHNOLOGY, INC. AND THE INDIVIDUAL CONTRIBUTORS
TO THE FAQ BY NECESSITY ASSUME NO RESPONSIBILITY FOR ACCURACY,
ERRORS OR OMISSIONS, OR FOR THE USES MADE OF ANY INFORMATION
AND/OR MATERIAL CONTAINED HEREIN OR ANY DECISION BASED ON SUCH
USE. NO WARRANTIES ARE MADE, EXPRESS OR IMPLIED, WITH REGARD TO
THE CONTENTS OF THIS WORK, ITS MERCHANTABILITY, OR FITNESS FOR A
PARTICULAR PURPOSE. BERKELEY DESIGN TECHNOLOGY, INC. AND THE
INDIVIDUAL CONTRIBUTORS SHALL NOT BE RESPONSIBLE FOR ANY DIRECT,
INDIRECT, SPECIAL, INCIDENTAL, OR CONSEQUENTIAL DAMAGES ARISING
OUT OF THE USE AND/OR RELIANCE ON THE CONTENTS OF THIS WORK.
Additionally, please note that the opinions expressed herein are
those of the individual contributors, and should not be construed
to be those of the contributor's employers or Berkeley Design
Q0.4: Redistribution Permission
This FAQ may be redistributed (in either electronic or printed
form) for non-commercial purposes provided that this notice is
preserved and that due credit is given to the maintainers and
Q0.5: Note on the list of manufacturers, addresses, and telephone numbers
The comp.dsp FAQ no longer includes a list of manufacturers. The
information becomes outdated in a few months, and we believe that
the list takes up an inappropriate amount of space in the FAQ
compared to the interest in the list.
Previous section (Overview) Next section (1)
Previous section (0) Next section (2)
Q1: General DSP
Q1.1: Summary of DSP books and significant research articles
Q1.1.1: Bibles of DSP theory
R. E. Crochiere and L. R. Rabiner, Multirate Digital Signal
Processing, Prentice-Hall, 1983, ISBN 0-13-605162-6.
This book is the only real reference for filter banks and
multirate systems, as opposed to being a tutorial.
Peter Kootsookos <p.kootsookos @mvt.ie=""> notes: 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. </p.kootsookos>Multirate Systems
and Filter Banks by P. P. Vaidyanathan covers this issue.
G. H. Golub and C. F. van Loan, Matrix Computations, Third
Edition, John Hopkins University Press, 1996, ISBN 081085413-X.
S. M. Kay, Modern Spectral Estimation: Theory and Application,
Prentice Hall, 1988, ISBN 0-13-598582-X.
R. Lyons, Understanding Digital Signal Processing, 2/E, Prentice
Hall Publishing Co., 2004, ISBN 0-13-108989-7.
Sanjit K. Mitra and James F. Kaiser, Handbook for Digital Signal
Processing, John Wiley and Sons, 1993, ISBN 0-471-61995-7.
Excellent reference work, but assumes you know a fair amount to
begin with. [Phil Lapsley]
A. V. Oppenheim, A. S. Willsky, and S. H. Nawab, Signals &
Systems, Prentice-Hall, Inc., 1996, ISBN 0-13-814757-4.
A. V. Oppenheim and R. W. Schafer, Digital Signal Processing,
Prentice-Hall, Inc., Englewood Cliffs, NJ, 1975, ISBN
A. V. Oppenheim and R. W. Schafer, Discrete-Time Signal
Processing, Prentice Hall, Englewood Cliffs, New Jersey 07632,
1989, ISBN 0-13-216292-X.
This is an updated version of the original, with some old
material deleted and lots of new material added.
S. J. Orfanidis, Optimum Signal Processing, Second Edition, 1989,
MacMillan Publishing, USA, ISBN 0-02-9498597.
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,
T.W. Parks and C. S. Burrus, DFT/FFT and Convolution Algorithms:
Theory and Implementation, John Wiley and Sons, 1985, ISBN
Thomas Parsons, Voice and Speech Processing, McGraw-Hill, 1987,
W. H. Press, S. A. Teukolsky, W. T. Vetterling, and B. P.
Flannery, Numerical Recipes in C, Second Edition, Cambridge
University Press, 1992, ISBN 0-52-143108-5.
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.
Q1.1.2: Adaptive signal processing
S. Haykin, Adaptive Filter Theory, 3rd Ed., Prentice Hall,
Englewood Cliffs, NJ, 1991. ISBN 0-13-322760-X.
J. R. Treichler, C. R. Johnson, and M. G. Lawrence, Theory and
Design of Adaptive Filters, John Wiley & Sons, New York, NY, 1987,
B. Widrow and S.D. Stearns, Adaptive Signal Processing,
Prentice-Hall, Inc., Englewood Cliffs, NJ, 1985. ISBN
Q1.1.3: Array signal processing
J.E. Hudson, Adaptive Array Principles, IEE London and New York,
Peter Peregrinus Ltd. Stevenage, UK and NY, 1981. ISBN
R.A. Monzingo and T.W. Miller, Introduction to Adaptive Arrays,
John Wiley and Sons, NY, 1980.
S. Haykin, J.H. Justice, N.L. Owsley, J.L. Yen, and A.C. Kak,
Array Signal Processing, Prentice-Hall, Inc., Englewood Cliffs,
D. H. Johnson and D. E. Dudgeon, Array Signal Processing, Concepts
and Techniques, Prentice-Hall, 1993. ISBN 0-13-048513-6.
R. T. Compton, Jr., Adaptive Antennas, Concepts and Performance,
Prentice-Hall, 1988, ISBN 0-13-004151-3.
Q1.1.4: Windowing articles
F. J. Harris, "On the Use of Windows for Harmonic Analysis with
the Discrete Fourier Transform", IEEE Proceedings, January 1978,
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
There are several typos in the above paper. The errors are
A. H. Nuttall, "Some Windows with Very Good Sidelobe Behavior,"
IEEE Trans. on Acoustics, Speech, and Signal Processing, Vol.
ASSP-29, No. 1, February 1981.
Nezih C. Geckinli and Davras Yavuz, "Some Novel Windows and a
Concise Tutorial Comparison of Window Families", IEEE Transactions
on Acoustics, Speech, and Signal Processing, Vol. ASSP-26, No. 6,
Lineu C. Barbosa, "A Maximum-Energy-Concentration Spectral
Window," IBM J. Res. Develop., Vol. 30, No. 3, May 1986, p.
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
D. J. Thomson, "Spectrum Estimation and Harmonic Analysis," Proc.
of the IEEE, vol. 70, no. 9, pp. 1055-1096, Sep. 1982.
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,
firstname.lastname@example.org, and Brian Evans,
Q1.1.5: Digital audio effects processing
Barry Blesser and J. Kates. "Digital Processing in Audio Signals."
in A. V. Oppenheim, ed., Applications of Digital Signal
Processing, Englewood Cliffs, NJ: Prentice-Hall, 1978. ISBN
Hal Chamberlin, Musical Applications of Microprocessors, 2nd Ed.,
Hayden Book Company, 1985.
Deta S. Davis, Computer Applications in Music: A Bibliography, 537
pages, ISBN 0-89579-225-7, pub: A-R Editions.
Charles Dodge and Thomas A. Jerse, Computer Music: Synthesis,
Composition, and Performance, NY: Schirmer Books, 1985. ISBN
Digital Signal Processing Committee of IEEE Acoustics, Speech, and
Signal Processing Society, ed., Programs for Digital Signal
Processing, New York: IEEE Press, 1979.
F. Richard Moore, Elements of Computer Music, Englewood Cliffs,
NJ: Prentice-Hall, 1990. ISBN: 0-13252-552-6.
Recommended. [Juhana Kouhia, email@example.com]
Ken C. Pohlmann, The Compact Disc: A Handbook of Theory and Use,
288 pages (cloth) ISBN 0-89579-234-6. (paper) ISBN 0-89579-228-1,
pub: A-R Editions.
Curtis Roads and John Strawn, ed., The Foundations of Computer
Music, Cambridge, MA: MIT Press, 1985.
Contains article on analysis/synthesis by Strawn, recommended;
also an another article maybe by J.A. Moorer [Juhana Kouhia,
Joseph Rothstein, Midi: A Comprehensive Introduction (Computer
Music and Digital Audio, Vol 7), 2nd Ed., A-R Editions, 1995. ISBN
Ken Steiglitz, A DSP Primer - With Applications to Digital Audio
and Computer Music, Addison-Wesley, 1996, 314 pp, softcover, ISBN
John Strawn, ed., Digital Audio Engineering, 144 pages, A-R
Editions. ISBN 0-86576-087-X.
John Strawn, ed., Digital Audio Signal Processing: An Anthology,
Los Altos, CA: W. Kaufmann, 1985. ISBN 0-86-576087-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,
John Strawn, ed., Digital Audio Signal Processing, 283 pages, ISBN
0-86576-082-9, pub: A-R Editions.
Recommended. [Quinn Jensen, firstname.lastname@example.org]
Curtis Roads, "A Computer Music History: Musical Automation from
Antiquity to the Computer Age"
David Cope, "Computer Analysis of Musical Style"
Dexter Morrill and Rick Taube, "A Little Book of Computer Music
James A. Moorer, About This Reverberation Business, Computer Music
Journal 3, 20 (1979): 13-28. (Also in Foundations of CM below).
Ok article, but you have to know basic DSP operations. [Juhana
Check more articles from Journal of the Audio Engineering Society
(JAES), for example more articles by Strawn.
[The above is largely from Quinn Jensen, email@example.com;
Juhana Kouhia, firstname.lastname@example.org; William Alves,
email@example.com; and Paul A Simoneau, firstname.lastname@example.org]
Q1.1.6: Digital signal processing implementation
User's manuals and data sheets on specific digital signal
processors are available directly from the manufacturers. The
works listed below may also be of interest.
A. Bateman and W. Yates, Digital Signal Processing Design,
Computer Science Press, MD, 1989.
R. Chassaing, Digital Signal Processing - Laboratory Experiments
Using C and the TMS320C31 DSK, Wiley, NY, ISBN 0-471-29362-8,
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,
Y. Dote, Servo Motor and Motion Control Using Digital Signal
Processors, Prentice Hall, NJ, 1990.
Mohamed El-Sharkawy, Digital Signal Processing Applications with
Motorola's 56002 Processor, Prentice Hall, Upper Sadle River, NJ,
ISBN 0-13-569476-0, 1996.
P. Embree, C Algorithms for Real-Time DSP, Prentice Hall,
Dale Grover and John R. Deller, Digital Signal Processing and the
Microcontroller, Prentice Hall, NJ, ISBN 0-13-081348-6, 1999.
J. L. Hennessy and D. A. Patterson, Computer Architecture: A
Quantitative Approach, Morgan Kaufmann Publishers, San Mateo, CA,
1990, ISBN 1-55-860329-8.
R. Higgins, Digital Signal Processing in VLSI, Prentice Hall, NJ,
1990. ISBN 0-13-212887-X.
It's a good primer on DSP theory and practice (albeit slightly
out of date regarding today's chips), aimed at both analog
engineers entering the digital realm and digital engineers
dealing with real-world problems. Its hardware orientation is
towards components and the Analog Devices ADSP-2100 series (just
emerging at the time of publication), but there is much in it of
fundamental tutorial value. [DanShein@ix.netcom.com]
B. A. Hutchins and T. W. Parks, A Digital Signal Processing
Laboratory Using the TMS320C25, Prentice Hall, NJ, 1990.
D. L. Jones and T. W. Parks, A Digital Signal Processing
Laboratory using the TMS32010, Prentice Hall, NJ, 1988.
N. Kehtarnavaz , Real-Time Digital Signal Processing : Based on
the TMS320C6000, Elsevier, 2004.Updated!
S. M. Kuo and B. H. Lee, Real-Time Digital Signal Processing:
Implementations, Application and Experiments with the TMS320C55x,
P. Lapsley, J. Bier, A. Shoham, and E. A. Lee, DSP Processor
Fundamentals: Architectures and Features, Berkeley Design
Technology, Inc., Fremont, CA, 1996.
Vijay Madisetti, VLSI Digital Signal Processors: An Introduction
to Rapid Prototyping and Design Synthesis, IEEE
Henrik V. Sorensen and Jianping Chen, A Digital Signal Processing
Laboratory Using the TMS320C30, Prentice Hall, Upper Sadle River,
NJ, ISBN 0-13-741828-0, 1997.
Steven A. Tretter, Communication system design using DSP
algorithms: with laboratory experiments for the TMS320C30, Plenum
Press, Norwell, MA, ISBN 0306450321, 1995.
S. A. Tretter, Communication system design using DSP algorithms:
with laboratory experiments for the TMS320C6700, Kluwer Academic
Q1.1.7: Free online books
The Scientist and Engineer's Guide to Digital Signal Processing
This introductory DSP book is available for free download at
http://www.dspguide.com/. Topics covered in this 640-page book
include: convolution, digital filters, audio processing, data
compression, and Fourier, Laplace, and z transforms.
Introduction to Statistical Signal Processing
This site provides the current version of the book Introduction
to Statistical Signal Processing by R.M. Gray and L.D. Davisson in
the Adobe portable document format (PDF). This format can be read
from a Web browser by using the Acrobat Reader helper application,
which is available at Adobe.
Yehar's Digital Sound Processing Tutorial for the Braindead
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
Q1.2: DSP training
Q1.2.1: Courses on DSP
DSP training is available from the following sources:
1. DSP Made Simple: basic DSP theory and algorithms. Web:
2. DSP without Tears: Z Domain Technologies covers theory and
applications. Web: http://www.dspwithouttears.com/
3. DSP Workshop: Dr. Bill Gordon, who is located in Austin,
gives them. He is a former Texas Instruments employee. He can
be reached at email@example.com. Web: http://www.dsp-workshops.com/
4. Berkeley Design Technology Inc.: BDTI is a DSP consulting and
independent DSP processor/tools evaluation firm in Berkeley,
CA. Web: http://www.bdti.com/
5. Cysip: Courses in DSP, Speech/Image Processing, and
Communications. Web: http://www.cysip.com/
[Brian Evans, firstname.lastname@example.org; Andreas Spanias,
Q1.2.2: On-Line courses on DSP
Updated Mar 1, 2003
Prof. Brian Evans: Real-time DSP course online at
TechOnLine (http://www.techonline.com/): Courses on various
Engineering Productivity Tools Ltd.
(http://www.eptools.com/tn/index.htm): Technical notes on various
topics (FFT, Sensor arrays, etc.).
BORES Signal Processing DSP course.
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
TI also has a site designed to help new DSP users (primarily new
TI DSP users) get started with their designs:
Q1.3: Where can I get free software for general DSP?
The packages listed below are mostly not oriented for use with a
specific DSP processor. See the later sections in the FAQ for
software relevant to a particular programmable DSP chip.
Q1.3.1: DSP Packages for MATLAB
FOR STUDENTS IN THE US AND CANADA: The MATLAB Student Version,
available from The MathWorks, is a full-featured version of MATLAB
and includes Simulink (with model sizes up to 300 blocks) and the
Symbolic Math toolbox. It is available for Windows and Linux. See
MATLAB user's group public domain extensions to MATLAB
The MATLAB Digest is issued at irregular intervals based on the
number of questions and software items contributed by users. To
subscribe to the newsletter, send mail to email@example.com.
To make submissions to the digest, please send to
firstname.lastname@example.org with a subject: "DIG" and description.
Some MATLAB tools are available on the web at
http://www.mathworks.com, or via anonymous ftp at
There is a set of Wavelet Tools available for MATLAB, see Section
2.9 of this FAQ.
We have developed a "Communications Toolbox" based on the MATLAB
code for classroom use. It is used by students taking a 4th year
communications course where the emphasis is on digital coding of
waveforms and on digital data transmission systems. The MATLAB
code that constitutes this toolbox has been in use for over two
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:
* Probability Theory
* Random Processes
* Binary Signalling Formats
* Digital Modulation
* Digital Communication
M-files (MATLAB 4.2) is available in:
The complete manual in Postscript format is available at
[Mehmet Zeytinoglu, email@example.com]
Digital Filter Package (DFP)
The Digital Filter Package is a GUI front-end to digital filter
design with MATLAB. DFP extends the basic digital filter design
functionality of MATLAB in two important ways:
* Filter coefficients can be quantized. This feature is
important if the filter is to be implemented on a fixed-point
* DFP generates assembly-language code for the designed digital
filter. In the current release of DFP, this option is only
available for the Motorola DSP56xxx family.
For more information:
Implementations of the CELP Federal Standard 1016 Speech Coder and LPC-10e
http://www.cysip.com/dsplinks.html. [Andreas Spanias,
Q1.3.2: DSP Packages for Mathematica
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.
Signal Processing Packages (SPP) and Notebooks, Version 2.9.5
Freely distributable extensions to Mathematica.
Enables the symbolic manipulation of signal
processing expressions: 1-D discrete/continuous
convolutions and 1-D/m-D linear transforms (Laplace,
Fourier, z, DTFT, and DFT). For linear transforms,
you can specify your own transform pairs and see the
intermediate computations. Great for showing students
how to take transforms, or for deriving input-output
relationships in a transform domain. Additional
abilities include analog filter design, solving DE's
using transforms, converting signal processing
expressions to their equivalent TeX forms, number
theoretic operations (Bezout numbers, Smith Form
decompositions, and matrix factors), and multirate
operations (graphical design of 2-d decimators).
Accompanying the SPPs are tutorial notebooks on
analog filter design, Fourier analysis, piecewise
convolution, and the z-transform (includes a
discussion of fundamentals of digital filter design).
These Notebooks illustrate difficult concepts (such
as the flip-and-slide view of convolution) through
Contact Brian Evans at firstname.lastname@example.org, or see
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
Dr. Roberto H. Bamberger reports: I have developed a
series of about 30 Lectures that I use for EE341
(Analog Communication Systems) here at Washington
State University. They use the SPP by Brian Evans.
They discuss many concepts associated with linear
systems theory. Topics covered include LTI system
theory, convolution, AM, FM, PM modulation and
demodulation, and the sampling theorem. NOTE: All
Notebooks were developed under NeXTSTEP 3.1 using
Mathematica 2.2. I make no guarantees about the
graphics being able to be rendered on anything other
than a NeXT.
Control Systems Analysis Package (COSYPAK) and Notebooks
Public domain extension to Mathematica. Classical and
state-space control analysis and design methods. The
Notebooks supplement the material in the textbook
"Modern Controls Theory" by Ogata. Largely based on
the Signal Processing Packages (SPP, see above).
For more information:
Contact Dr. Sreenath, email@example.com.
Other Mathematica DSP Notebooks
The following Mathematica notebooks can be ftped from
* pub/malcolm/FilterDesign.math IIR Filter Design
(continuous and discrete)
* pub/malcolm/ear.math.Z Implementation of Lyon's
* pub/malcolm/Gammatone.math Implementation of
Gammatone Cochlear Model. Printed copies (with
floppies) are available from the Apple library
(firstname.lastname@example.org). Pointers to the
notebooks are available from Malcolm Slaney's
The following Mathematica notebooks (from Julius
Smith, email@example.com) can be ftped from
* pub/DSP/Tutorials/GenHamming.ma.Z Generalized
* pub/DSP/Tutorials/Kaiser.ma.Z The Kaiser window
* pub/DSP/Tutorials/WinFlt.ma.Z Digital filter
design by the "window method"
(There are other DSP related items in pub/DSP on
ccrma-ftp; see other sections of this FAQ for
Q1.3.3: Other DSP Libraries
Audio File I/O Routines
The Audio File Signal Processing (AFsp) package is a
library of routines for reading and writing audio
files of various formats. It also provides utility
programs for comparing audio files (speech activity
factor, SNR); coping, combining, concatenating, and
changing the format of audio files; resampling
(arbitrary sample rate conversion); filtering audio
files (including ITU-T filters); and generating noise
/ tones. These routines are freely distributable
under a license similar to the GNU license. They were
written by Prof. Peter Kabal of the
Telecommunications and Signal Processing Library at
The kit is located at:
For more information:
[Brian Evans, firstname.lastname@example.org]
FFTW ("Fastest Fourier Transform in the West")
FFTW, a fast C FFT library, along with benchmarks
comparing the speed and accuracy of many public
domain FFTs on a variety of platforms.
For more information:
Intel Signal Processing Library
The Intel Signal Processing Library provides a set of
optimized C functions that implement typical signal
processing operations on Intel processors.
ISIP Automatic Speech Recognition System
Source code for a public domain automatic speech
ISIP Foundation Classes
A large C++ class library for use in signal
processing research. Includes classes for file I/O,
vector and matrix operations, signal processing,
pattern recognition, and automatic speech
Linear Systems Toolbox for Maple
Public domain extension to Maple.
For more information:
Contact Tony Richardson, email@example.com.
Signal Processing using C++ (SPUC)
Free C++ classes for DSP & digital communications
simulation and modeling. Includes:
* Basic building blocks such as fixed bit width
integer classes, pure-delay blocks, Gaussian and
random noise, etc.
* DSP building blocks such as FIR, IIR, Allpass,
Running Average, Lagrange interpolation filters,
NCOs (numerically controlled oscillators),
* Several communications functions such as timing,
phase and frequency discriminators for BPSK/QPSK
signals and raised-cosine type FIR filter
For more information:
Vector/Signal/Image Processing Library (VSIPL)
VSIPL is an API and library for vector, signal, and
Q1.3.4: DSP Software
The AudioFile System (AF) is a device-independent
network-transparent audio server. The distribution
includes device drivers and server code for Digital
RISC systems running Ultrix, Digital Alpha AXP
systems running OSF/1, and Sun Microsystems
SPARCstations running SunOS. Also included are an API
and library, out-of-the-box core applications, and a
number of contributed applications. AudioFile allows
applications to generate and process audio in
real-time and at present handles up to 48 KHz stereo
AudioFile is distributed in source form, with a
copyright allowing unrestricted use for any purpose
except sale (see the Copyright notice).
The kit is located in the at:
A sample kit of sound-bites is available as:
For more information:
firstname.lastname@example.org is a mailing list for discussions of
AudioFile. Send mail to email@example.com to be
added to this list. [Larry Stewart,
VisiQuest (previously known as Khoros Pro)
Visual programming interface for image and video
processing. See the UseNet group
comp.soft-sys.khoros. VisiQuest is a commercial
product, but free licenses are available to students
using the product in a profit-free manner. For more
A variety of Unix platforms, Windows 2000 and Windows
XP, Mac OS X. (Note that the native Windows versions
are scheduled for release in January 2005.)
VisiQuest can be obtained from the AccuSoft website:
MathViews, WaveXplorer, MathXplorer
MathViews for Windows/32 - Math Software for Windows
3.1 (version 2.1 only) and Windows 95/NT. Current
version is 2.21. "MathViews for Windows/32 is MATLAB
look-alike. It has a full set of linear algebra and
signal processing functionality. MathViews is highly
compatible with the MATLAB language"
WaveXplorer for Windows 95/NT: version 2.21.
"Interactive waveform editor (based on the
computational engine of MathViews)"
MathXplorer, MathViews ActiveX control: version 2.21.
"MathXplorer provides easy access to the MathViews
computational engine that can be embedded in MS
Excel, Visual Basic, Internet Explorer, etc."
Author: Dr. Shalom Halevy, firstname.lastname@example.org,
PO BOX 22564, San Diego, CA 92192 (619) 552-9031 USA
http://www.mathwizards.com/. No sources. Shareware
P.C. convolution is a educational software package
that graphically demonstrates the convolution
operation. It runs on IBM PC type computers using DOS
4.0 or later. It is currently being used in schools
of Mathematics, Electrical Engineering, Earth
Sciences, Aeronautics, Astronomy, Geophysics, and
The current version of this software demonstrates
continuous time convolution, discrete time, and
circular convolution along with cross-correlation.
instructors may obtain a free, fully operational
version by contacting Dr. Kurt Kosbar at the address
Dr. Kurt Kosbar
117 Electrical Engineering Building
University of Missouri - Rolla
Rolla, Missouri, USA 65401, phone: (573) 341-4894
Ptolemy is an object oriented framework for the
specification, simulation, and rapid prototyping of
systems. From a flow graph description, Ptolemy can
generate both C code and DSP assembly code for rapid
prototyping. Code generation is not yet complete and
is included in the current release for demonstration
Ptolemy is available for Solaris, HPUX, Digital Unix,
Linux, and Windows NT.
Ptolemy is available via anonymous ftp. Get the file:
ftp://ptolemy.eecs.berkeley.edu/pub/README and follow
Organizations without Internet access can obtain
Ptolemy, without support, from ILP. This is often a
more stable, less featured version than is available
EECS/ERL Industrial Liaison Program Office
205 Cory Hall
University of California, Berkeley
Berkeley, CA 94720
This includes printed documentation, including
installation instructions, a user's guide, and manual
pages. A handling fee will be charged.
For more information about Ptolemy and its successor, Ptolemy
See http://ptolemy.eecs.berkeley.edu and the
comp.soft-sys.ptolemy Usenet newsgroup.
SANTIS (now Dataplore)
SANTIS is a tool for Signal ANalysis and TIme Series
processing. All operations can be executed from a
mouse-supported graphical user interface. It contains
standard facilities for signal processing as well as
advanced features like wavelet techniques and methods
of nonlinear dynamics.
Supported systems include Microsoft Windows, Linux,
Solaris, and SGI Irix.
You can get the software and more information from
the WWW page http://datan.de/dataplore/. [Ralf
ScopeDSP is a time and frequency signal processing
tool for Windows 95/NT. It can read and or write real
or complex, time or frequency sampled data in a
variety of file formats. It can generate various
types of time signals, manipulate data, and transform
between time and frequency domains. Shareware with a
60-day test period.
Sfront is a compiler for Structured Audio, the audio
signal processing language that is a part of the
ISO/IEC MPEG 4 Audio standard. The output of the
compiler is a C program, that when compiled and
executed generates the audio, with many audio input,
audio output, and control options, including
real-time interactive and audio streaming support for
some OS's. The website also includes an online book
for learning how to program in Structured Audio, and
a reference manual that describes how to extend
sfront and embed it in applications.
The compiler is written in strict ANSI C, and runs on
most UNIX systems as well as MS Windows.
Sfront is distributed under the GNU public license,
and is available for free download at the website:
Shorten is a compressor/coder for waveform files. It
supports both lossless coding and lossy coding down
to three bits per sample. It operates using a linear
predictor and Huffman coding the prediction residual
using Rice codes. A technical report shows that this
simple scheme is both fast and near optimal. Data
formats supported are RIFF WAVE plus signed and
unsigned values at 8 or 16 bits per sample, ulaw,
alaw and multiple interleaved channels. For lossless
compression of speech files recorded using 16 bits at
16 kHz the compression ratio is typically 2:1. CD
audio (44.1 kHz, 16 bit stereo) is near transparant
at 4:1 or 5:1 lossy compression.
The command line version compiles on most UNIX
platforms. A version is available for MS Windows/NT.
http://www.softsound.com/Shorten.html points to all
versions. [Tony Robinson, email@example.com]
Q1.3.5: Text to Speech Conversion Software
Free (but not public domain) text to speech
conversion software is available via anonymous ftp
from wilma.cs.brown.edu in the pub directory as
speak.tar.Z. It will compile and run on a SPARC's
built-in audio after modifying speak.c with the path
of your libaudio.h (e.g.,
/usr/demo/SOUND/libaudio.h). It's a simple phoneme
concatenation system with commensurate synthesized
speech quality (a directory of phoneme audio files is
included). [Joe Campbell, firstname.lastname@example.org]
A public domain version of the same Naval Research
Lab text to phoneme rules can be obtained from:
The comp.speech FTP site includes a 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, email@example.com]
Q1.3.6: Filter Design Software
Updated Sep 9 2004
* There are many filter design programs available via anonymous
FTP or by HTTP. The following are summarized here and
discussed in greater detail below:
* August 1992 IEEE Trans. on Signal Processing: METEOR FIR
filter design program.
* DFiltFIR and DFiltInt FIR filter design program.
* Netlib IIR filter design.
* IEEE Press "Programs for Digital Signal Processing".
* Tod Schuck's near-optimal Kaiser-Bessel program.
* Brian Evans' and Niranjan Damera-Venkata's packages for
Matlab and Mathematica.
* Charles Poynton's filter design resource page.
* Juhana Kouhia's hotlist.
* Alex Matulich's recipes for compiling 2-pole digital
* The August 92 issue of IEEE Transactions on Signal Processing
includes a paper entitled "METEOR: A Constraint-Based FIR
Filter Design Program" by Kenneth Steiglitz, Thomas W. Parks
and James F. Kaiser. The authors describe an FIR design
program which allows specification of the target frequency
response characteristics in a fairly generalized and flexible
way. As well as designing filters, the program can optimize
filter lengths and push band limits.
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
There is no manual. The paper includes instructions on
running the programs. [Steve Clift, firstname.lastname@example.org]
Weimin Liu has created a Windows 95 interface to the Meteor
program, which can be downloaded from
* Other free filter design packages are DFiltFIR and DFiltInt.
DFiltFIR designs minimax approximation FIR filters. It uses
the algorithm developed by McClelland and Parks and
incorporates constraints on the response as proposed by
Grenez. DFiltInt designs minimum mean-square error FIR
interpolating filters. The design specification is in terms
of a tabulated power spectrum model for the input signal.
The packages are available from
or directly via anonymous ftp from
Another package, libtsp, is a library of C-language routines
for signal processing. The package is available from
or directly via anonymous ftp from
ftp://ftp.tsp.ece.mcgill.ca/pub/libtsp/ [Peter Kabal,
* Another source is netlib: "A free program to design IIR
Butterworth, Chebyshev, and Cauer (elliptic) filters, in any
of lowpass, bandpass, band reject, and high pass
configurations, is available in netlib (e.g.,
netlib.bell-labs.com) as the file netlib/cephes/ellf.shar.Z.
By email to email@example.com the request message
text is `send ellf from cephes'. The URL is
http://www.netlib.org. [Stephen Moshier,
* The Fortran source code from the IEEE Press book "Programs
For Digital Signal Processing" is available by anonymous ftp
from ftp://soma.crl.mcmaster.ca/pub/IEEE/software/dsp.zip or
includes FIR and IIR filter design software, FFT subroutines,
interpolation programs, a coherence and cross-spectral
estimation program, linear prediction analysis programs, and
a frequency domain filtering program. There is also a C/C++
version of the McClellan-Parks-Rabiner FIR filter design
program available from
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 firstname.lastname@example.org; also Andrew
* I have developed a MATLAB (vers 4.0 for Windows) program that
allows for the frequency domain design of the "near optimal"
Kaiser-Bessel window. The program is based upon the three
closed form equations developed by Kaiser and Schafer in 1981
that allow for the specification of the time domain window
length, and the frequency domain mainlobe width and relative
sidelobe amplitude. For signal processing applications where
the spectral content of the windowing function is critical so
as not to mask adjacent spectra such as radar signal
processing applications where a weak target return adjacent
to a strong target return could be easily masked by a
windowing function that resolves poorly in frequency; this
program allows complete frequency domain specification of the
spectral characteristics of the windowing function. The
current version of this program allows for the user to
specify the two frequency domain parameters of mainlobe width
and relative sidelobe amplitude and lets the window length
fall out as the dependent variable. The program is easily
modified to allow for any two parameters to be selected and
allowing the third to be determined as a result.
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
* Filter Optimization Packages for Matlab and Mathematica,
version 1.1 by Brian L. Evans and Niranjan Damera-Venkata,
Dept. of ECE, The University of Texas at Austin. Available
We have released a set of Matlab packages to optimize the
following characteristics of analog filter designs
1. magnitude response
2. linear phase in the passband
3. peak overshoot in the step response
4. quality factors (Q)
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 . An earlier version of the
framework is described in . We plan to extend this
framework to digital IIR filters.
 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.
 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.
[Brian Evans, email@example.com]
* ScopeFIR is a FIR filter design tool for Windows 95/NT which
designs complex FIR filters using the Parks-McClellan
algorithm or windowing. It can then mix, scale, quantize, and
edit the FIR coefficients. It creates a wide variety of
impulse and frequency response plots, and supports many data
file formats, including TI assembly and ADI PM. Shareware
with a 60-day trial period, available from
[Grant Griffin, firstname.lastname@example.org]
* FilterExpress is a free filter synthesis tool for Windows. It
supports the design and analysis of IIR, FIR and multirate
FIR filters. It is available for download from
* DSP Design Performance provides Java applets generating
different filters. The applets can be found at
* Charles Poynton has an extensive list of hot-links to filter
design resources on the web at
* Juhana Kouhia has an extensive list of links at
* Alex Matulich has compiled recipes (step by step
instructions) for coding three kinds of 2-pole digital
filters, both low-pass and high-pass, complete with
correction factors to ensure that the 3 dB cutoff frequency
stays where you put it when you cascade filters of the same
Alex has made these recipes available here:
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.
Q1.3.7: Audio effects
Harmony Central publishes some of the source code for its
synthesis and audio processing program at
http://www.harmony-central.com/Computer/Programming/. The code may
be used in public releases, but Harmony Central asks you to credit
the author and possibly make the product available for free or
publish any modified code.
Music-DSP Source Code Archive
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, email@example.com]
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