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• FAQ 1 General DSP
• FAQ 2 - Algorithms and standards
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Q1: General 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://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.

{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]

Q1.2: DSP training

Besser Associates (http://www.bessercourse.com/): Basic DSP theory and algorithms.

Z Domain Technologies (http://www.zdt.com/~dsp): DSP theory and applications.

DSP Workshops (http://www.dsp-workshops.com/): DSP system design.

Berkeley Design Technology, Inc. (http://www.bdti.com): DSP processors.

[Brian Evans, bevans@combo.ece.utexas.edu]

Prof. Brian Evans: Real-time DSP course online at http://www.ece.utexas.edu/~bevans/courses/realtime/.

TechOnLine (http://www.techonline.com/): Courses on various topics.

[Brian Evans, bevans@combo.ece.utexas.edu]

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.

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: ftp://ftp.mathworks.com/pub/contrib/v4/misc/comm_tbx/

The complete manual in Postscript format is available at ftp://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]

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: ftp://mathsource.wri.com/pub/NumberedItems/0204-297-0011
Windows: ftp://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).

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.

 

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: ftp://crl.dec.com/pub/DEC/AF/

A sample kit of sound-bites is available as: ftp://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.

 

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]

 

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]

 

Linear Systems Toolbox for Maple

Description:

Public domain extension to Maple.

To obtain:

ftp://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/.

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:

ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/syntheses/english2phoneme.tar.gz

The comp.speech FTP site includes a speech synthesis directory at ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis. The main package is "rsynth" which is a complete text to speech synthesis system. Several component packages are also present. "textnorm" converts non-words such as digit strings into words (e.g. 1000 to ONE THOUSAND). "english2phoneme" does some of the same but its main functionality is to guess an appropriate phoneme sequence for each word. "klatt" takes a parametric form that describes each phoneme and converts it to a waveform. Other packages exist in the same directory to edit and visualise the klatt parameters. [Tony Robinson, ajr@softsound.com]

• The 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 ftp://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_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]

• 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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