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Q1: General DSP
Q1.1: Summary of DSP books and significant research articles
Updated 6/3/98
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 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, 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, 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://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,
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 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, 1999.
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.
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,
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: DSP training
Updated 11/25/99
Q1.2.1: Courses on DSP
-
DSP training is available from the following sources:
-
DSP Made Simple: basic DSP theory and algorithms. Web: http://www.bessercourse.com/
-
DSP without Tears: Z Domain Technologies covers theory and applications.
Web: http://www.zdt.com/~dsp/
-
DSP Workshop: Dr. Bill Gordon, who is located in Austin, gives them. He
is a former Texas Instruments employee. He can be reached at dsp@io.com.
Web: http://www.dsp-workshops.com/
-
Berkeley Design Technology Inc.: BDTI is a DSP consulting and independent
DSP processor/tools evaluation firm in Berkeley, CA. Web: http://www.bdti.com/
-
Cysip: Courses in DSP, Speech/Image Processing, and Communications. Web:
http://www.cysip.com/
[Brian Evans, bevans@combo.ece.utexas.edu; Andreas Spanias,
spanias@asu.edu]
Q1.2.2: On-line courses on DSP
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.
[Brian Evans, bevans@combo.ece.utexas.edu]
Q1.3: 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.
Q1.3.1: DSP Packages for MATLAB
Updated 11/18/99
-
FOR STUDENTS: Prentice Hall has published a Student Edition of MATLAB
for PCs and Macs. The software is limited in matrix size (128 x 128 matrix;
16,384 elements). It includes the Signal Processing, Control System, and
Symbolic Math toolboxes.
Windows version (MATLAB 5.3): ISBN 0-13-022598-3
Macintosh version (MATLAB 5.0): ISBN 0-13-272485-5
For general info: matlab@prenhall.com (or http://www.mathworks.com/products/studentedition/).
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 http://www.mathworks.com/products/studentversion/.
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]
Implementation of the CELP Federal Standard 1016 Speech Coder
-
To obtain:
-
http://www.cysip.com/dsplinks.html.
[Andreas
Spanias, spanias@asu.edu]
GSM Routines
-
Description:
-
Chris Stratford has placed GSM-related MATLAB code online, including
routines for GMSK modulation and Viterbi equalization.
-
To obtain:
-
http://www.stratfordc.free-online.co.uk.
Q1.3.2: 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:
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).
Q1.3.3: Other DSP Libraries
Updated 10/18/99
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]
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://www.fftw.org
-
For more information:
-
fftw@fftw.org.
Intel Signal Processing Library
-
Description:
-
The Intel Signal Processing Library provides a set of optimized C
functions that implement typical signal processing operations on Intel
processors.
-
To obtain:
-
http://developer.intel.com/vtune/perflibst/SPL/
ISIP Automatic Speech Recognition System
-
Description:
-
Source code for a public domain automatic speech recognition system.
-
To obtain:
-
http://www.isip.msstate.edu/projects/speech/software/asr/index.html
ISIP Foundation Classes
-
Description:
-
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 recognition.
-
To obtain:
-
http://www.isip.msstate.edu/projects/speech/education/tutorials/isip_env/
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.
Signal Processing using C++ (SPUC)
-
Description:
-
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), Cordic
rotator.
-
Several communications functions such as timing, phase and frequency discriminators
for BPSK/QPSK signals and raised-cosine type FIR filter functions.
-
To obtain:
-
http://spuc.webjump.com
-
For more information:
-
tony_kirke@ieee.org.
Vector/Signal/Image Processing Library (VSIPL)
-
Description:
-
VSIPL is an API and library for vector, signal, and image processing.
-
To obtain:
-
http://www.vsipl.org
Q1.3.4: DSP Software
Updated 10/18/99
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]
Khoros
-
Description:
-
Visual programming interface for image and video processing. See the UseNet
group
comp.soft-sys.khoros. A free
trial version is available.
-
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/.
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.
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
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 Solaris, HPUX, Digital Unix, Linux, and Windows
NT.
-
To Obtain:
-
Ptolemy is available via anonymous ftp. Get the file:
ftp://ptolemy.eecs.berkeley.edu/pub/README
and follow the instructions.
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.
-
For more information about Ptolemy and its successor, Ptolemy II:
-
See http://ptolemy.eecs.berkeley.edu
and the
comp.soft-sys.ptolemy
Usenet newsgroup.
SANTIS (now Dataplore)
-
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, Solaris, and SGI Irix.
-
To obtain:
-
You can get the software and more information from the WWW page
http://datan.de/dataplore/.
[Ralf
Vandenhouten, vanni@Physiology.RWTH-Aachen.DE]
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/.
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 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.
-
Platforms:
-
The command line version compiles on most UNIX platforms. A version is
available for MS Windows/NT.
-
To obtain:
-
http://www.softsound.com/Shorten.html
points to all versions.
[Tony Robinson, ajr@softsound.com]
Q1.3.5: 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/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]
Q1.3.6: Filter Design Software
Updated 9/2/99
-
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.
-
Brian Evans' and Niranjan Damera-Venkata's packages for Matlab and Mathematica.
-
ScopeFIR.
-
FilterExpress.
-
Charles Poynton'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.
-
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:
-
magnitude response
-
linear phase in the passband
-
peak overshoot in the step response
-
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_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/papers/filter_design_ecctd95.ps.Z.
[Brian Evans, bevans@combo.ece.utexas.edu]
-
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 http://www.iowegian.com/scopefir.htm.
[Grant Griffin, grant.griffin@iowegian.com]
-
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
http://www.systolix.co.uk/swdownload.htm.
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