comp.dsp FAQ [1 of 4]
Archive-name: dsp-faq/part1
Last-modified: Wed September 30 1998
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: September 30, 1998
- Ole Wolf, FAQ maintainer
_________________________________________________________________
0. WHAT IS COMP.DSP?
0.1 DISCLAIMER OF WARRANTY
0.2 Redistribution permission
0.3 Note on the list of manufacturers, addresses, and telephone
numbers
1. GENERAL DSP
1.1 DSP book and article references
1.1.1 Bibles of DSP
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 chips
1.2 Where can I get free software for general DSP?
1.2.1 DSP packages for MATLAB
1.2.2 DSP packages for Mathematica
1.2.3 Other DSP software
1.2.4 Text to Speech Conversion Software
1.2.5 Filter design software
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 is S/PDIF?
2.6.1 Where can I get copies of ITU (formerly CCITT)
standards?
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
information?
2.9.2 What are some good books and papers on wavelets?
2.9.3 Where can I get some software for wavelets?
2.10 How do I calculate the coefficients for a Hilbert transformer?
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 an assembler for the ADSP-2105?
3.3.2 Where can I get algorithms or libraries for Analog
Devices DSPs?
3.4 Software for Lucent Technologies (Formerly AT&T) DSPs
3.5 Software for Motorola DSPs
3.5.1 Where can I get a free assembler for the Motorola
DSP56000?
3.5.2 Where can I get a free C compiler for the Motorola
DSP56000?
3.5.3 Where can I get algorithms and libraries for
Motorola DSPs?
3.5.4 Where can I get NeXT-compatible Motorola DSP56001
code?
3.5.5 Where can I get emulators for the 68HC11 (6811)
processor?
3.6 Software for Texas Instruments DSPs
3.6.1 Where can I get free algorithms or libraries for TI
DSPs? What is the number for the TI DSP BBS?
3.6.2 Where can I get a free C compiler for the TI
TMS320C30?
3.6.3 Where can I get a free assembler for the TI
TMS320C30?
3.6.4 Where can I get a free simulator for the TI
TMS320C30?
3.6.5 What is Tick? Where can I get it?
4. DSP DEVELOPMENT BOARDS
5. OPERATING SYSTEMS
People involved...
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.
Other relevant news groups are:
+ comp.arch.embedded
+ comp.compression
+ comp.realtime
+ comp.speech
+ sci.i>
Transfer interrupted!
Relevant FAQs are:
+ Higher-order statistics FAQ
+ comp.arch.embedded FAQ
+ comp.compression FAQ
+ comp.realtime FAQ
+ comp.speech FAQ
+ sci.image.processing FAQ
+ Audio sampling FAQ
Other relevant links:
+ http://weblab.research.att.com/phoaks/comp/dsp/
+ http://www.ncc.com/misc/dsp_sites.html
+ http://www.tasp.org
+ http://www.eg3.com/dsp/index.htm,
http://www.cera2.com/dsp/index.htm, or
http://www.eetoolbox.com/dsp/index.htm
+ http://www.iowegian.com/splat.htm
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Q0.1: 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 Technology, Inc.
Phew.
Q0.2: 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
contributors.
Q0.3: Note on the list of manufacturers, addresses, and telephone numbers
In the latest revision of the comp.dsp FAQ, we decided to
eliminate the long list of manufacturers. The information
becomes outdated in a few months, and we believe that the list
takes up an inappropriate amount of space of 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
Updated 6/3/98
Q1.1.1: BIBLES OF DSP
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
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. 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. G. Lyons, Understanding Digital Signal Processing,
Addison-Wesley Publishing Co., 1997. ISBN 0-201-63467-8.
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 and R. W. Schafer, Digital Signal Processing,
Prentice-Hall, Inc., Englewood Cliffs, N.J., 1975. ISBN
0-13-214635-5.
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,
jkahrs@castor.atlas.de]
T.W. Parks and C. S. Burrus, DFT/FFT and Convolution
Algorithms: Theory and Implementation, John Wiley and Sons,
1985, ISBN 0-47-181932-8.
Thomas Parsons, Voice and Speech Processing, McGraw-Hill,
1987, ISBN 0-07-048541-0.
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://cfatab.harvard.edu/nr/nronline.html
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, ISBN 0-47-183220-0.
B. Widrow and S.D. Stearns, Adaptive Signal Processing,
Prentice-Hall, Inc., Englewood Cliffs, N.J., 1985. ISBN
0-13-004029-0
_________________________________________________________________
Q1.1.3: ARRAY SIGNAL PROCESSING
J.E. Hudson, Adaptive Array Principles, IEE London and New
York, Peter Peregrinus Ltd. Stevenage, U.K., and New York,
1981. ISBN 0-86-341143-6.
R.A. Monzingo and T.W. Miller, Introduction to Adaptive
Arrays, John Wiley and Sons, New York, 1980.
S. Haykin, J.H. Justice, N.L. Owsley, J.L. Yen, and A.C. Kak,
Array Signal Processing, Prentice-Hall, Inc., Englewood
Cliffs, N.J., 1985.
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, pp. 51-83.
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:
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, December 1978.
Lineu C. Barbosa, "A Maximum-Energy-Concentration Spectral
Window," IBM J. Res. Develop., Vol. 30, No. 3, May 1986, p.
321-325.
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]
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,
wei@vision.ece.utexas.edu, and Brian Evans, bevans@ece.utexas.edu]
_________________________________________________________________
Q1.1.5: DIGITAL AUDIO EFFECTS PROCESSING
Books:
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 0-13-039115-8.
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, New York: Schirmer Books, 1985.
ISBN 0-02-873100-X.
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, jk87377@cc.tut.fi]
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,
jk87377@cc.tut.fi]
Joseph Rothstein, Midi: A Comprehensive Introduction (Computer
Music and Digital Audio, Vol 7), 2nd Ed., A-R Editions, 1995.
ISBN 0-89-579309-1.
Ken Steiglitz, A DSP Primer - With Applications to Digital
Audio and Computer Music, Addison-Wesley, 1996, 314 pp,
softcover, ISBN 0-8053-1684-1.
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,
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]
Forthcoming books:
{please let us know at comp-dsp-faq@bdti.com if they are out!}
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
Instruments"
Articles:
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
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
]
_________________________________________________________________
Q1.1.6: DIGITAL SIGNAL PROCESSING CHIPS
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 with C and the
TMS320C30, Wiley, N. Y., 1992.
R. Chassaing and D. W. Horning, Digital Signal Processing with
the TMS320C25, Wiley, N. Y., 1990.
Y. Dote, Servo Motor and Motion Control Using Digital Signal
Processors, Prentice Hall, N. J. , 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.
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,
N. J., 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, N. J., 1990.
D. L. Jones and T. W. Parks, A Digital Signal Processing
Laboratory using the TMS32010, Prentice Hall, N. J., 1988.
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
Press/Butterworth-Heinemann, 1995.
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.
_________________________________________________________________
Q1.2: Where can I get free software for general DSP?
Updated 6/3/98
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.
Q1.2.1: DSP PACKAGES FOR MATLAB
Updated 12/31/96
Note: FOR STUDENTS: Prentice Hall has published a student
edition of MATLAB which contains a book and set of disks for
PCs and Macs. The software is limited only in matrix size (32
x 32 matrix; 1024 elements) and in its ability to import or
call C or Fortran subroutines. On the plus side, it is able to
run without a coprocessor (it will use one if it is present)
and it includes a subset of the Signal Processing and Controls
Toolboxes, The Signals and Systems Toolbox, which provides for
added functionality.
Book only: ISBN =0-13-856006-4;
Book + disk: ISBN=0-13-855974-0 for 3.5" or ISBN=0-13-855982-1
for 5.25"
Macintosh version: ISBN=0-13-855990-2.
For general info: matlab@prenhall.com. [From the MATLAB Users
Group (Editor, hwilson@ua1vm.ua.edu)]
MATLAB user's group public domain extensions to MATLAB
Description:
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
subscribe@mathworks.com. To make submissions to the digest,
please send to hwilson@ua1vm.ua.edu with a subject: "DIG" and
description.
To obtain:
Some MATLAB tools are available on the web at
http://www.mathworks.com, or via anonymous ftp at
ftp://ftp.mathworks.com/.
Wavelet Tools
Description:
There is a set of Wavelet Tools available for MATLAB, see
Section 2.9 of this FAQ.
Communications Toolbox
Description:
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 years.
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
+ Quantization
+ Binary Signalling Formats
+ Detection
+ Digital Modulation
+ Digital Communication
To obtain:
M-files (MATLAB 4.2) is available in:
file://ftp.mathworks.com/pub/contrib/v4/misc/comm_tbx/
The complete manual in Postscript format is available at
file://ftp.mathworks.com/pub/contrib/v4/misc/comm_tbx.manual.
ps. [Mehmet Zeytinoglu, mzeytin@ee.ryerson.ca]
Digital Filter Package (DFP)
Description:
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
DSP processor.
+ 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:
http://www.ee.ryerson.ca:8080/~mzeytin/dfp/index.html. [Mehmet
Zeytinoglu, mzeytin@ee.ryerson.ca]
_________________________________________________________________
Q1.2.2: DSP PACKAGES FOR MATHEMATICA
Updated 1/13/97
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
Description:
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 animation.
To obtain:
ftp to ftp.eedsp.gatech.edu/Mathematica.
A freely distributable Notebook reader is available for
Macintosh computers and IBM-compatibles running MicroSoft
Windows by anonymous ftp: Mac:
file://mathsource.wri.com/pub/NumberedItems/0204-297-0011
Windows:
file://mathsource.wri.com/pub/NumberedItems/0203-599-0011
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.
For more information:
Contact Brian Evans at bevans@ece.utexas.edu, or see
http://www.ece.utexas.edu/~bevans/projects/spp.html.
EE341
Description:
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
Description:
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).
To obtain:
anonymous ftp veda.esys.cwru.edu (129.22.40.9).
For more information:
Contact Dr. Sreenath, sree@veda.esys.cwru.edu.
Other Mathematica DSP Notebooks
The following Mathematica notebooks can be ftped from
worldserver.com:
+ pub/malcolm/FilterDesign.math IIR Filter Design (continuous
and discrete)
+ pub/malcolm/ear.math.Z Implementation of Lyon's Cochlear
Model
+ pub/malcolm/Gammatone.math Implementation of Gammatone
Cochlear Model. Printed copies (with floppies) are available
from the Apple library (corp.lib@applelink.apple.com).
Pointers to the notebooks are available from Malcolm Slaney's
homepage at http://www.interval.com/~malcolm/pubs.html.
The following Mathematica notebooks (from Julius Smith,
jos@ccrma.stanford.edu) can be ftped from
ccrma-ftp.stanford.edu:
+ pub/DSP/Tutorials/GenHamming.ma.Z Generalized Hamming windows
+ 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 details).
_________________________________________________________________
Q1.2.3: OTHER DSP SOFTWARE
Updated 1/7/97
Ptolemy
Description:
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 purposes only.
Platforms:
Ptolemy is available for the Sun 4 (sparc), DecStation (MIPS),
and HP (HP-PA) architectures. Installing the system requires
90 Mbytes for Ptolemy (more if you optionally remake). Ptolemy
also requires at least 8 Mbytes of physical memory. Linux
binaries also exist.
To Obtain:
Ptolemy is available via anonymous ftp. Get the file:
file://ptolemy.eecs.berkeley.edu/pub/README and follow the
instructions. ptolemy.eecs.berkeley.edu contains the entire
Ptolemy distribution, a postscript version of the Ptolemy
manual, and several Ptolemy papers.
Organizations without Internet FTP capability 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.
For more information
See the comp.soft-sys.ptolemy Usenet newsgroup.
Khoros
Description:
Visual programming interface for image and video processing.
See the UseNet group comp.soft-sys.khoros.
Platforms:
Digital UNIX 4.0D, Red Hat Linux 4.2, Irix 6.2 and 6.3, Solaris
2.5.1, Windows NT 4.0
To obtain:
Khoros is found at: http://www.khoral.com/.
PC Convolution
Description:
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
Experimental Psychology.
The current version of this software demonstrates continuous
time convolution, discrete time, and circular convolution along
with cross-correlation.
To obtain:
ftp://lamarr.ee.umr.edu/pub/pcc5.zip. University instructors
may obtain a free, fully operational version by contacting Dr.
Kurt Kosbar at the address listed below.
Dr. Kurt Kosbar
117 Electrical Engineering Building
University of Missouri - Rolla
Rolla, Missouri, USA 65401, phone: (314) 341-4894
e-mail: kk@ee.umr.edu
AudioFile System
Description:
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 audio.
To obtain:
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: file://crl.dec.com/pub/DEC/AF/
A sample kit of sound-bites is available as:
file://crl.dec.com/pub/DEC/AF/AF2R2-other.tar
For more information:
af@crl.dec.com is a mailing list for discussions of AudioFile.
Send mail to af-request@crl.dec.com to be added to this list.
[Larry Stewart, stewart@crl.dec.com]
Audio File I/O Routines
Description:
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 copying,
comparing, filtering, resampling, and playback of audio files.
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 McGill
University.
To obtain:
The kit is located at: ftp://ftp.TSP.EE.McGill.CA/pub/AFsp/.
For more information:
See http://www.TSP.EE.McGill.CA/software/AFsp/AFsp.html. [Brian
Evans, bevans@combo.ece.utexas.edu]
MathViews, WaveXplorer, MathXplorer
Description:
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, shalevy@mathwizards.com, PO BOX
22564, San Diego, CA 92192 (619) 552-9031 USA (Tel/FAX)
http://www.mathwizards.com.
To obtain:
http://www.mathwizards.com/. No sources. Shareware version
available.
Shorten
Description:
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 simple linear predictor and Huffman
coding the prediction residual using Rice codes. A technical
report included with the distribution shows that this simple
scheme is both fast and near optimal. Data formats supported
are 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) sounds pretty good at 4:1 or 5:1 lossy compression.
Platforms:
The code compiles on most (all known) UNIX platforms and an
MS-DOS executables also provided. Version 2.1 was released on
26 Feb 1996.
To obtain:
ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/shorten.tar.
gz. [Tony Robinson, ajr@softsound.com]
SANTIS
Description:
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.
Platforms:
Supported systems include Microsoft Windows, Linux, SUN Solaris
2.3, and SGI Irix 5.3.
To obtain:
You can get the software and more information from the WWW page
http://www.physiology.rwth-aachen.de/bs/signal/santis. The
software is also available by anonymous ftp from
ftp://ftp.physiology.rwth-aachen.de/pub/santis. [Ralf
Vandenhouten, vanni@Physiology.RWTH-Aachen.DE]
Linear Systems Toolbox for Maple
Description:
Public domain extension to Maple.
To obtain:
file://ftp.egr.duke.edu/pub/maple/linsys1.2.tar.Z
For more information:
Contact Tony Richardson, amr@mpl.ucsd.edu.
FFTW ("Fastest Fourier Transform in the West")
Description:
FFTW, a fast C FFT library, along with benchmarks comparing the
speed and accuracy of many public domain FFTs on a variety of
platforms.
To obtain:
http://theory.lcs.mit.edu/~fftw
For more information:
fftw@theory.lcs.mit.edu.
ScopeDSP
Description:
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.
To obtain:
http://www.iowegian.com/.
_________________________________________________________________
Q1.2.4: TEXT TO SPEECH CONVERSION SOFTWARE
Updated 1/7/97
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, jpcampb@afterlife.ncsc.mil]
A public domain version of the same Naval Research Lab text to
phoneme rules can be obtained from:
file://svr-ftp.eng.cam.ac.uk/pub/comp.speech/syntheses/english
2phoneme.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]
_________________________________________________________________
Q1.2.5: FILTER DESIGN SOFTWARE
Updated 9/30/98
* There are filter design programs available via anonymous FTP. 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
+ Charles Poyton's filter design resource page
+ Juhana Kouhia's hotlist.
* 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 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.
* 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
http://www.TSP.EE.McGill.CA/software/FilterDesign/ or directly
via anonymous ftp from
ftp://ftp.TSP.EE.McGill.CA/pub/FilterDesign.
Another package, libtsp, is a library of C-language routines for
signal processing. The package is available from
http://www.TSP.EE.McGill.CA/software/libtsp/ or directly via
anonymous ftp from ftp://ftp.TSP.EE.McGill.CA/pub/libtsp. [Peter
Kabal, kabal@tsp.ee.mcgill.ca]
* 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 netlib@netlib.bell-labs.com
the request message text is `send ellf from cephes'. The URL is
http://www.netlib.org. [Stephen Moshier, moshier@world.std.com]
* 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
ftp://soma.crl.mcmaster.ca/pub/IEEE/software/dsp.tar.gz. It
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
file://ftp.uu.net/usenet/comp.sources.misc/volume22/fir/part01.Z
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,
Witold.Waldman@dsto.defence.gov.au, from Charles Owen at
mgcbo@uxa.ecn.bgu.au; also Andrew Ukrainec, ukrainec@InfoUkes.com]
* 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
NAWCAD Patuxent River
Combat Identification Section
Code 4.5.8.2.3.1
St. Inigoes, MD 20684-0010
e-mail: tod_schuck@idsmail.combat-edt.navy.mil
* 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 from
http://www.ece.utexas.edu/~bevans/projects/syn_filter_software.
html.
We have released a set of Matlab packages to optimize the
following characteristics of analog filter designs simultaneously:
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 [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_optimizatio
n/.
[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/papers/filter_design_ecctd95.ps
.Z.
[Brian Evans, bevans@combo.ece.utexas.edu]
* Charles Poynton has an extensive list of hot-links to filter
design resources on the web at
http://www.inforamp.net/~poynton/Poynton-dsp.html.
* Juhana Kouhia has an extensive list of links at
http://www.funet.fi/~kouhia/hotlist-dsp.html.
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