Archive-name: dsp-faq/part1 Last-modified: Thu May 4 2006 URL: http://www.bdti.com/faq/ 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: May 4, 2006 - Kenton Williston, FAQ maintainer ---------------------------------------------------------------------- 0. What is comp.dsp? 0.1 Relevant links 0.2 Versions of the comp.dsp FAQ 0.3 DISCLAIMER OF WARRANTY 0.4 Redistribution permission 0.5 Note on the list of manufacturers, addresses, and telephone numbers 1. General DSP 1.1 DSP book and article references 1.1.1 Bibles of DSP theory 1.1.2 Adaptive signal processing 1.1.3 Array signal processing 1.1.4 Windowing articles 1.1.5 Digital audio effects processing 1.1.6 Digital signal processing implementation 1.1.7 Free online books 1.2 DSP training 1.2.1 Courses on DSP 1.2.2 On-Line courses on DSP 1.3 Where can I get free software for general DSP? 1.3.1 DSP packages for MATLAB 1.3.2 DSP packages for Mathematica 1.3.3 Other DSP libraries 1.3.4 DSP software 1.3.5 Text to Speech Conversion Software 1.3.6 Filter design software 1.3.7 Audio effects 2. Algorithms and standards 2.1 Where can I get public domain algorithms for DSP? 2.2 What are CELP and LPC? Where can I get source for them? 2.3 What is ADPCM? Where can I get source for it? 2.4 What is GSM? Where can I get source for it? 2.5 How does pitch perception work, and how do I implement it? 2.6 What standards exist for digital audio? What is AES/EBU? What 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? 2.11 Algorithm implementation: floating-point versus fixed-point 3. Programmable DSP chips and their software 3.1 What are the available DSP chips and chip architectures? 3.2 What is the difference between a DSP and a microprocessor? 3.3 Software for Analog Devices DSPs 3.3.1 Where can I get a C compiler for the ADSP-21xx and ADSP-21xxx? 3.3.2 Where can I get tools for the ADSP-21xxx? 3.3.3 Where can I get an assembler for the ADSP-2105? 3.3.4 Where can I get algorithms or libraries for Analog Devices DSPs? 3.4 Software for Agere Systems (Formerly Lucent Technologies) DSPs 3.5 Software for Motorola DSPs 3.5.1 Where can I get a free assembler for the Motorola DSP56000? 3.5.2 Where can I get a free C compiler for the Motorola DSP56000? 3.5.3 Where can I get a disassembler for the Motorola DSP56000? 3.5.4 Where can I get algorithms and libraries for Motorola DSPs? 3.5.5 Where can I get NeXT-compatible Motorola DSP56001 code? 3.5.6 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? 3.6.2 Where can I get free development tools for TI DSPs? 3.6.3 Where can I get a free C compiler for the TI TMS320C3x/4x? 3.6.4 Where can I get a free assembler for the TI TMS320C3x/4x? 3.6.5 Where can I get a free simulator for the TI TMS320C3x/4x? 3.6.6 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. Q0.0: Original Charter Contributed by Max Hauser: Fifteen years ago, comp.dsp sprang into life. The group's original charter is presented below. From: todd@ivucsb.sba.ca.us (Todd Day) Newsgroups: comp.dsp Subject: Welcome! Message-ID: <1989Sep20.195449.3833@ivucsb.sba.ca.us> Date: 20 Sep 89 19:54:49 GMT Reply-To: todd@ivucsb.sba.ca.us (Todd Day) Organization: Disillusioned Graduate Hackers, Santa Barbara, CA Lines: 34 Just so people know what this new group is all about, I am reposting the proposal for this group: DSP is an acronym for Digital Signal Processing. DSP is currently a rapidly growing field. Most people have come in contact with DSP either through compact disc players or satellite photos. Here are the proposed topics to be covered by this group: 1) Discussions of DSP hardware a) single processors b) DSP boards for computers c) new DSP product announcements d) architecture 2) Discussions of DSP software a) source code listings for particular chip b) how to use development software on particular chip c) general purpose DSP software on computers 3) Discussions of DSP theory a) general algorithms b) other devices that use DSP (CD players, etc.) 4) Discussions of DSP applications a) audio b) image c) control d) communications e) speech f) etc. -- Todd Day | todd@ivucsb.sba.ca.us | ivucsb!todd@anise.acc.com "Just give me a killer sound system and the babes will follow." Q0.1: Relevant links Other relevant news groups are: * comp.arch.embedded * comp.compression * comp.realtime * comp.speech.research * sci.image.processing 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 There is an index of DSP-related mailing lists at: * http://www.dsprelated.com/ Other relevant links: * http://www.eg3.com/dsp/index.htm, http://www.cera2.com/dsp/index.htm, or http://www.eetoolbox.com/dsp/index.htm * http://www.dspguru.com * http://shoko.calarts.edu/~glmrboy/musicdsp/music-dsp.html Q0.2: Versions of the comp.dsp FAQ If you're reading this via the World Wide Web: Click on http://www.bdti.com/faq/dsp_faq.zip or http://www.bdti.com/faq/dsp_faq.tar.Z to download a compressed HTML version of the FAQ. Click on http://www.bdti.com/faq/dsp_faq.asc.zip or http://www.bdti.com/faq/dsp_faq.asc.tar.Z to download a compressed ASCII version of the FAQ. (When you click on these links, your browser should tell you that it can't display the files and then ask you if you want to download them instead. Say "yes.") If you're reading this as ASCII text: Get with the program and get a web browser. The FAQ is available on World Wide Web with a much nicer interface. This is especially true for information presented in tabular form. Try: http://www.bdti.com/faq Q0.3: DISCLAIMER OF WARRANTY BERKELEY DESIGN TECHNOLOGY, INC. AND THE INDIVIDUAL CONTRIBUTORS TO THE FAQ BY NECESSITY ASSUME NO RESPONSIBILITY FOR ACCURACY, ERRORS OR OMISSIONS, OR FOR THE USES MADE OF ANY INFORMATION AND/OR MATERIAL CONTAINED HEREIN OR ANY DECISION BASED ON SUCH USE. NO WARRANTIES ARE MADE, EXPRESS OR IMPLIED, WITH REGARD TO THE CONTENTS OF THIS WORK, ITS MERCHANTABILITY, OR FITNESS FOR A PARTICULAR PURPOSE. BERKELEY DESIGN TECHNOLOGY, INC. AND THE INDIVIDUAL CONTRIBUTORS SHALL NOT BE RESPONSIBLE FOR ANY DIRECT, INDIRECT, SPECIAL, INCIDENTAL, OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE AND/OR RELIANCE ON THE CONTENTS OF THIS WORK. Additionally, please note that the opinions expressed herein are those of the individual contributors, and should not be construed to be those of the contributor's employers or Berkeley Design Technology, Inc. Phew. Q0.4: Redistribution Permission This FAQ may be redistributed (in either electronic or printed form) for non-commercial purposes provided that this notice is preserved and that due credit is given to the maintainers and contributors. Q0.5: Note on the list of manufacturers, addresses, and telephone numbers The comp.dsp FAQ no longer includes a list of manufacturers. The information becomes outdated in a few months, and we believe that the list takes up an inappropriate amount of space in the FAQ compared to the interest in the list. Previous section (Overview) Next section (1) Previous section (0) Next section (2) Q1: General DSP Q1.1: Summary of DSP books and significant research articles Updated 12/17/01 Q1.1.1: Bibles of DSP theory R. E. Crochiere and L. R. Rabiner, Multirate Digital Signal Processing, Prentice-Hall, 1983, ISBN 0-13-605162-6. This book is the only real reference for filter banks and multirate systems, as opposed to being a tutorial. Peter Kootsookos <p.kootsookos@mvt.ie> notes: this book is most certainly an excellent book on multi-rate signal processing, but it came out right before perfect reconstruction filter banks hit the streets. 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, NJ, 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, NJ, 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, UK and NY, 1981. ISBN 0-86-341143-6. R.A. Monzingo and T.W. Miller, Introduction to Adaptive Arrays, John Wiley and Sons, NY, 1980. S. Haykin, J.H. Justice, N.L. Owsley, J.L. Yen, and A.C. Kak, Array Signal Processing, Prentice-Hall, Inc., Englewood Cliffs, NJ, 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, NY: 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, NY, 1992. R. Chassaing and D. W. Horning, Digital Signal Processing with the TMS320C25, Wiley, NY, 1990. R. Chassaing, DSP Applications Using C and the TMS320C6x DSK, Wiley, NY, ISBN 0471207543, 2002. J. Datta, B. Karley, J. Lane, and J. Norwood, DSP Filter Cookbook, Prompt, 2000.Updated! Y. Dote, Servo Motor and Motion Control Using Digital Signal Processors, Prentice Hall, NJ, 1990. Mohamed El-Sharkawy, Digital Signal Processing Applications with Motorola's 56002 Processor, Prentice Hall, Upper Sadle River, NJ, ISBN 0-13-569476-0, 1996. P. Embree, C Algorithms for Real-Time DSP, Prentice Hall, 1995.Updated! Dale Grover and John R. Deller, Digital Signal Processing and the Microcontroller, Prentice Hall, NJ, ISBN 0-13-081348-6, 1999. J. L. Hennessy and D. A. Patterson, Computer Architecture: A Quantitative Approach, Morgan Kaufmann Publishers, San Mateo, CA, 1990, ISBN 1-55-860329-8. R. Higgins, Digital Signal Processing in VLSI, Prentice Hall, NJ, 1990. ISBN 0-13-212887-X. It's a good primer on DSP theory and practice (albeit slightly out of date regarding today's chips), aimed at both analog engineers entering the digital realm and digital engineers dealing with real-world problems. Its hardware orientation is towards components and the Analog Devices ADSP-2100 series (just emerging at the time of publication), but there is much in it of fundamental tutorial value. [DanShein@ix.netcom.com] B. A. Hutchins and T. W. Parks, A Digital Signal Processing Laboratory Using the TMS320C25, Prentice Hall, NJ, 1990. D. L. Jones and T. W. Parks, A Digital Signal Processing Laboratory using the TMS32010, Prentice Hall, NJ, 1988. N. Kehtarnavaz , Real-Time Digital Signal Processing : Based on the TMS320C6000, Elsevier, 2004.Updated! S. M. Kuo and B. H. Lee, Real-Time Digital Signal Processing: Implementations, Application and Experiments with the TMS320C55x, Wiley, 2001.Updated! 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. S. A. Tretter, Communication system design using DSP algorithms: with laboratory experiments for the TMS320C6700, Kluwer Academic Publishers, 2003.Updated! ---------------------------------------------------------------------- Q1.1.7: Free online books Updated 2/11/02 The Scientist and Engineer's Guide to Digital Signal Processing This introductory DSP book is available for free download at http://www.dspguide.com/. Topics covered in this 640-page book include: convolution, digital filters, audio processing, data compression, and Fourier, Laplace, and z transforms. Yehar's sound DSP tutorial for the braindead This tutorial is for people with "high school level" math knowledge, so you won't have to be a specialized genius to be able to read this. There's actually quite a lot information in this one, but the best covered subjects are: filters in general, FIR and IIR filter design, interpolation, frequency shifting. http://www.student.oulu.fi/~oniemita/DSP/INDEX.HTM [Steve Horne, steve@lurking.demon.co.uk] ---------------------------------------------------------------------- Q1.2: DSP training Updated 05/06/02 Q1.2.1: Courses on DSP DSP training is available from the following sources: 1. DSP Made Simple: basic DSP theory and algorithms. Web: http://www.bessercourse.com/ 2. DSP without Tears: Z Domain Technologies covers theory and applications. Web: http://www.zdt.com/ 3. DSP Workshop: Dr. Bill Gordon, who is located in Austin, gives them. He is a former Texas Instruments employee. He can be reached at dsp@io.com. Web: http://www.dsp-workshops.com/ 4. Berkeley Design Technology Inc.: BDTI is a DSP consulting and independent DSP processor/tools evaluation firm in Berkeley, CA. Web: http://www.bdti.com/ 5. Cysip: Courses in DSP, Speech/Image Processing, and Communications. Web: http://www.cysip.com/ [Brian Evans, bevans@ece.utexas.edu; Andreas Spanias, spanias@asu.edu] ---------------------------------------------------------------------- Q1.2.2: On-Line courses on DSP Updated Mar 1, 2003 Prof. Brian Evans: Real-time DSP course online at http://www.ece.utexas.edu/~bevans/courses/realtime/. TechOnLine (http://www.techonline.com/): Courses on various topics. Engineering Productivity Tools Ltd. (http://www.eptools.com/tn/index.htm): Technical notes on various topics (FFT, Sensor arrays, etc.). BORES Signal Processing DSP course. (http://www.bores.com/courses/intro/index.htm): Introduction courses to DSP. TI has a centralized training site where DSP designers can access all of TI's training webcasts, workshops and seminars. It can be found at www.dspvillage.ti.com/trainingpr2. It covers TI DSP, tools, software and applications. Analog training is also included. TI also has a site designed to help new DSP users (primarily new TI DSP users) get started with their designs: http://www.dspvillage.ti.com/cocostu. ---------------------------------------------------------------------- Q1.3: Where can I get free software for general DSP? Updated 05/06/02 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 05/06/02 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: 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/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] Implementations of the CELP Federal Standard 1016 Speech Coder and LPC-10e 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 04/03/01 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: Contact Brian Evans at bevans@ece.utexas.edu, or see http://www.ece.utexas.edu/~bevans/projects/symbolic/spp.html. 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. 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). 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). ---------------------------------------------------------------------- Q1.3.3: Other DSP Libraries Updated 05/06/02 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 comparing audio files (speech activity factor, SNR); coping, combining, concatenating, and changing the format of audio files; resampling (arbitrary sample rate conversion); filtering audio files (including ITU-T filters); and generating noise / tones. These routines are freely distributable under a license similar to the GNU license. They were written by Prof. Peter Kabal of the Telecommunications and Signal Processing Library at McGill University. To obtain: The kit is located at: ftp://ftp.tsp.ece.mcgill.ca/TSP/AFsp/ For more information: See http://www.tsp.ece.mcgill.ca/Docs/Software/AFsp/AFsp.html [Brian Evans, bevans@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/software/products/perflib/spl/index.htm 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/software/documentation/class/index.html 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. 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.sourceforge.net/ 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: 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] VisiQuest (previously known as Khoros Pro) Description: Visual programming interface for image and video processing. See the UseNet group comp.soft-sys.khoros. VisiQuest is a commercial product, but free licenses are available to students using the product in a profit-free manner. For more information, see http://www.accusoft.com/imaging/visiquest/students.asp. Platforms: A variety of Unix platforms, Windows 2000 and Windows XP, Mac OS X. (Note that the native Windows versions are scheduled for release in January 2005.) To obtain: VisiQuest can be obtained from the AccuSoft website: http://www.accusoft.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: (573) 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/. Sfront Description: Sfront is a compiler for Structured Audio, the audio signal processing language that is a part of the ISO/IEC MPEG 4 Audio standard. The output of the compiler is a C program, that when compiled and executed generates the audio, with many audio input, audio output, and control options, including real-time interactive and audio streaming support for some OS's. The website also includes an online book for learning how to program in Structured Audio, and a reference manual that describes how to extend sfront and embed it in applications. Platforms: The compiler is written in strict ANSI C, and runs on most UNIX systems as well as MS Windows. To obtain: Sfront is distributed under the GNU public license, and is available for free download at the website: http://www.cs.berkeley.edu/~lazzaro/sa. 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: 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] ---------------------------------------------------------------------- Q1.3.6: Filter Design Software Updated Sep 9 2004 * There are many filter design programs available via anonymous FTP or by HTTP. The following are summarized here and discussed in greater detail below: * August 1992 IEEE Trans. on Signal Processing: METEOR FIR filter design program. * DFiltFIR and DFiltInt FIR filter design program. * Netlib IIR filter design. * IEEE Press "Programs for Digital Signal Processing". * Tod Schuck's near-optimal Kaiser-Bessel program. * Brian Evans' and Niranjan Damera-Venkata's packages for Matlab and Mathematica. * ScopeFIR. * FilterExpress. * Charles Poynton's filter design resource page. * Juhana Kouhia's hotlist. * Alex Matulich's recipes for compiling 2-pole digital filters. * 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.ece.mcgill.ca/Docs/Software/FilterDesign/FilterDesign.html or directly via anonymous ftp from ftp://ftp.tsp.ece.mcgill.ca/TSP/FilterDesign/. Another package, libtsp, is a library of C-language routines for signal processing. The package is available from http://www.tsp.ece.mcgill.ca/reports/Software/libtsp/libtsp.html or directly via anonymous ftp from ftp://ftp.tsp.ece.mcgill.ca/pub/libtsp/ [Peter Kabal, kabal@ECE.McGill.CA] * 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, 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 Lockheed Martin NE&SS Moorestown, NJ 08060 e-mail: tod.m.schuck@lmco.com * Filter Optimization Packages for Matlab and Mathematica, version 1.1 by Brian L. Evans and Niranjan Damera-Venkata, Dept. of ECE, The University of Texas at Austin. Available from http://www.ece.utexas.edu/~bevans/projects/filters/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/publications/papers/95/filter_design_ecctd95/ [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. * DSP Design Performance provides Java applets generating different filters. The applets can be found at http://www.nauticom.net/www/jdtaft. * 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. * Alex Matulich has compiled recipes (step by step instructions) for coding three kinds of 2-pole digital filters, both low-pass and high-pass, complete with correction factors to ensure that the 3 dB cutoff frequency stays where you put it when you cascade filters of the same type together. Alex has made these recipes available here: http://unicorn.us.com/alex/2polefilters.html The recipes cover Butterworth, Critically-Damped,

# comp.dsp FAQ [1 of 4]

Started by ●May 4, 2006

Reply by ●May 4, 20062006-05-04

Archive-name: dsp-faq/part2 Last-modified: Thu May 4 2006 URL: http://www.bdti.com/faq/ Previous section (1) Next section (3) Q2: Algorithms and standards Q2.1: Where can I get public domain algorithms for general-purpose DSP? Updated 12/31/96 The following archives contain things such as matrix operations, FFT's and generally useful things like that, as opposed to complete applications. Netlib Netlib serves some of this software via email. Try mail to netlib@ORNL.GOV with "send help" in the subject field. To Obtain: For Europe: Internet: netlib@nac.no EARN/BITNET: netlib%nac.no@norunix.bitnet X.400: s=netlib; o=nac; c=no; EUNET/uucp: nac!netlib For more information: See Jack J. Dongarra and Eric Grosse, "Distribution of Mathematical Software Via Electronic Mail," Comm. ACM (1987) 30,403--407. A similar collection of statistical software is available from statlib@temper.stat.cmu.edu. The symbolic algebra system REDUCE is supported by reduce-netlib@rand.org. NSWC Library The Naval Surface Warfare Center has a library of mathematical Fortran subroutines that may be of use. The NSWC library is a library of general-purpose Fortran subroutines that provide a basic computational capability in a variety of mathematical activities. Emphasis has been placed on the transportability of the codes. Subroutines are available in the following areas: Elementary Operations, Geometry, Special Functions, Polynomials, Vectors, Matrices, Large Dense Systems of Linear Equations, Banded Matrices, Sparse Matrices, Eigenvalues and Eigenvectors, l1 Solution of Linear Equations, Least-Squares Solution of Linear Equations, Optimization, Transforms, Approximation of Functions, Curve Fitting, Surface Fitting, Manifold Fitting, Numerical Integration, Integral Equations, Ordinary Differential Equations, Partial Differential Equations For more information: NSWC Library of Mathematical Subroutines Report No.: NSWC TR 90-21, January 1990 by Alfred H. Morris, Jr. Naval Surface Warfare Center (E43) Dahlgren, VA 22448-5000 U.S.A. [Witold Waldman] IEEE Press book "Programs For Digital Signal Processing" You can get the Fortran source code from the IEEE Press book "Programs For Digital Signal Processing." See question 1.3.6. ---------------------------------------------------------------------- Q2.2: What are CELP and LPC? Where can I get the source for CELP and LPC? Updated 09/10/01 CELP stands for "code excited linear prediction". LPC stands for "linear predictive coding". They are compression algorithms used for low bit rate (2400 and 4800 bps) speech coding. The U.S. DoD's Federal-Standard-1016 based 4800 bps code excited linear prediction voice coder version 3.2 (CELP 3.2) Fortran and C simulation source codes are available for worldwide distribution (on DOS diskettes, but configured to compile on Sun SPARC stations) from NTIS and DTIC. Example input and processed speech files are included. A Technical Information Bulletin (TIB), "Details to Assist in Implementation of Federal Standard 1016 CELP," and the official standard, "Federal Standard 1016, Telecommunications: Analog to Digital Conversion of Radio Voice by 4,800 bit/second Code Excited Linear Prediction (CELP)," are also available. To obtain CELP: Available through the National Technical Information Service: NTIS U.S. Department of Commerce 5285 Port Royal Road Springfield, VA 22161 USA (800) 553-6847 FS-1016 CELP 3.2 may also be obtained from ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/celp_3.2a.tar.Z. LPC-10 (2.4 Kbps) is available from ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/lpc10-1.0.tar.gz. LPC (4.8 Kbps) can be downloaded in SpeakFreely http://www.speakfreely.org/, or in HawkVoice http://www.hawksoft.com/hawkvoice/. HawkVoice includes versions of OpenLPC, LPC-10, LPC, GSM, and Intel/DVI ADPCM. These versions have been rewritten to support multiple encoding and decoding streams, and the interfaces have been standardized. [Phil Frisbie, Jr., phil@hawksoft.com] OpenLPC (1.4 and 1.8 Kbps) can be downloaded from ftp://ftp.futuredynamics.com/OpenLPC/. MATLAB software for LPC-10 is available from http://www.eas.asu.edu/~spanias/srtcrs.html. Also, postscript copies of tutorials of speech coding can be found at http://www.eas.asu.edu/~spanias/papers.html. [Andreas Spanias, spanias@asu.edu] For more information: * The following articles describe the Federal-Standard-1016 4.8-kbps CELP coder (it's unnecessary to read more than one): Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, The Federal Standard 1016 4800 bps CELP Voice Coder, Digital Signal Processing, Academic Press, 1991, Vol. 1, No. 3, p. 145-155. Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, The DoD 4.8 kbps Standard (Proposed Federal Standard 1016), in Advances in Speech Coding, ed. Atal, Cuperman and Gersho, Kluwer Academic Publishers, 1991, Chapter 12, p. 121-133. Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, The Proposed Federal Standard 1016 4800 bps Voice Coder: CELP, Speech Technology Magazine, April/May 1990, p. 58-64. Additional information on CELP can also be found in the comp.speech FAQ. * The voicing classifier used in the enhanced LPC-10 (LPC-10e) is described in: Campbell, Joseph P., Jr. and T. E. Tremain, Voiced/Unvoiced Classification of Speech with Applications to the U.S. Government LPC-10E Algorithm, Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, 1986, p. 473-6. The U. S. Federal Standard 1015 (NATO STANAG 4198) is described in: Thomas E. Tremain, The Government Standard Linear Predictive Coding Algorithm: LPC-10, Speech Technology Magazine, April 1982, pp. 40-49. [Most of the above from Joe Campbell, jpcampb@afterlife.ncsc.mil, with additions from Dan Frankowski, drankow@winternet.com, and Ed Hall, edhall@rand.org] ---------------------------------------------------------------------- Q2.3: What is ADPCM? Where can I get source for it? Updated: 04/03/01 ADPCM stands for Adaptive Differential Pulse Code Modulation. It is a family of speech compression and decompression algorithms. A common implementation takes 16-bit linear PCM samples and converts them to 4-bit samples, yielding a compression rate of 4:1. To obtain: There is public domain C code available via anonymous ftp at ftp://ftp.cwi.nl/pub/audio/adpcm.shar written by Jack Jansen (email Jack.Jansen@cwi.nl). It is very programmer-friendly. The ADPCM code used is the Intel/DVI ADPCM code which is being recommended by the IMA Digital Audio Technical Working Group. It allows the following calls: adpcm_coder(short inbuf[], char outbuf[], int nsample, struct adpcm_state *state); adpcm_decoder(char inbuf[], short outbuf[], int nsample, struct adpcm_state *state); Note that this is NOT a G.722 coder. The ADPCM standard is much more complicated, probably resulting in better quality sound but also in much more computational overhead. Platforms: The routines have been tested on numerous platforms, and will easily compress and decompress millions of samples per second on current hardware. For more information: The G.721/722/723 packages are available from ITU at http://www.itu.ch/. This is also available as: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/G711_G722_G723.tar.gz [From Dan Frankowski, dfrankow@winternet.com; Jack Jansen, Jack.Jansen@cwi.nl] ---------------------------------------------------------------------- Q2.4: What is GSM? Where can I get source for it? Updated 4/27/00 GSM (Global System for Mobile Communication) is a standard for digital cellular telephony used in Europe. GSM also refers to the speech coder used in GSM telephones, which is what this section of the FAQ is concerned with. The Communications and Operating Systems Research Group (KBS) at the Technische Universitaet Berlin is currently working on a set of UNIX-based tools for computer-mediated telecooperation that will be made freely available. As part of this effort they are publishing an implementation of the European GSM 06.10 provisional standard for full-rate speech transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse excitation/long term prediction) coding at 13 kbit/s. GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility with typical UNIX applications, our implementation turns frames of 160 16-bit linear samples into 33-byte frames (1650 Bytes/s). The quality of the algorithm is good enough for reliable speaker recognition; even music often survives transcoding in recognizable form (given the bandwidth limitations of 8 kHz sampling rate). The interfaces offered are a front end modeled after compress(1), and a library API. Compression and decompression run faster than real time on most SPARCstations. The implementation has been verified against the ETSI standard test patterns. Jutta Degener jutta@cs.tu-berlin.de, Carsten Bormann cabo@cs.tu-berlin.de) Communications and Operating Systems Research Group, TU Berlin Fax: +49.30.31425156, Phone: +49.30.31424315 To obtain: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/gsm-1.0.6.tar.gz. An alternative site is ftp://ftp.cwi.nl/pub/audio/gsm-1.0.7.tar.gz. Try also: http://kbs.cs.tu-berlin.de/~jutta/toast.html. [From Dan Frankowski, dfrankow@winternet.com; Jutta Degener, jutta@cs.tu-berlin.de] ---------------------------------------------------------------------- Q2.5: How does pitch perception work, and how do I implement it on my DSP chip? Updated 04/02/01 Pitch is officially defined as "That attribute of auditory sensation in terms of which sounds may be ordered on a musical scale." Several good examples illustrating the subtleties of pitch perception are included in the "Auditory Demonstrations CD" which is available from the Acoustical Society of America, Woodbury, NY 10797 for $20. Books/papers: A good general reference about the psychology of pitch perception is the book: B.C.J. Moore, An Introduction to the Psychology of Hearing, Academic Press, London, 1997. This book is available in paperback and makes a good desk reference. An algorithm implementation that matches a large body of psycho-acoustical work, but which is computationally very intensive, is presented in the paper: Malcolm Slaney and Richard Lyon, "A Perceptual Pitch Detector," Proceedings of the International Conference of Acoustics, Speech, and Signal Processing, 1990, Albuquerque, New Mexico. Available for ftp at ftp://worldserver.com/pub/malcolm/ICASSP90.psc.Z The definitive papers describing the use of such a perceptual pitch detector as applied to the classical pitch literature is in: Ray Meddis and M. J. Hewitt. "Virtual pitch and phase sensitivity of a computer model of the auditory periphery. " Journal of the Acoustical Society of America 89 (6 1991): 2866-2682. and 2883-2894. The current work that argues for a pure spectral method starts with the work of Goldstein: J. Goldstein, "An optimum processor theory for the central formation of the pitch of complex tones," Journal of the Acoustical Society of America 54, 1496-1516, 1973. Two approaches are worth considering if something approximating pitch is appropriate. The people at IRCAM have proposed a harmonic analysis approach that can be implemented on a DSP: Boris Doval and Xavier Rodet, "Estimation of Fundamental Frequency of Musical Sound Signals," Proceedings of the 1991 International Conference on Acoustics, Speech, and Signal Processing, Toronto, Volume 5, pp. 3657-3660. The classic paper for time domain (peak picking) pitch algorithms is: B. Gold and L. Rabiner, "Parallel processing techniques for estimating pitch periods of speech in the time domain," Journal of the Acoustical Society of America, 46, pp 441-448, 1969. Finally, a word of caution: Pitch is not single-valued. We can hear a sound and match it to several different pitches. Imagine the number of instruments in an orchestra, each with its own pitch. Even a single sound can have more than one pitch. See for example Demonstration 27 from the ASA Auditory Demonstrations CD. [The above from Malcolm Slaney, Interval Research, and John Lazzaro, U.C. Berkeley.] Information about independently changing the pitch and speed of a digital recording can be found at http://www.dspdimension.com/html/timepitch.html. [Stephan M. Bernsee, spam@dspdimension.com]Updated! ---------------------------------------------------------------------- Q2.6: What standards exist for digital audio? What is AES/EBU? What is S/PDIF? Updates 1/8/97 Q2.6.1: Where can I get copies of ITU (formerly CCITT) standards? Try the ITU (International Telecommunication Union) homepage at http://www.itu.ch/. ---------------------------------------------------------------------- Q2.6.2: What standards are there for digital audio? AES/EBU The "AES/EBU" (Audio Engineering Society / European Broadcast Union) digital audio standard is probably the most popular digital audio standard today. Most consumer and professional digital audio devices (CD players, DAT decks, etc.) that feature digital audio I/O support AES/EBU. AES/EBU is a bit-serial communications protocol for transmitting digital audio data through a single transmission line. It provides two channels of audio data (up to 24 bits per sample), a method for communication control and status information ("channel status bits"), and some error detection capabilities. Clocking information (i.e., sample rate) is derived from the AES/EBU bit stream, and is thus controlled by the transmitter. The standard mandates use of 32 kHz, 44.1 kHz, or 48 kHz sample rates, but some interfaces can be made to work at other sample rates. AES/EBU provides both "professional" and "consumer" modes. The big difference is in the format of the channel status bits mentioned above. The professional mode bits include alphanumeric channel origin and destination data, time of day codes, sample number codes, word length, and other goodies. The consumer mode bits have much less information, but do include information on copy protection (naturally). Additionally, the standard provides for "user data", which is a bit stream containing user-defined (i.e., manufacturer-defined) data. According to Tim Channon, "CD user data is almost raw CD subcode; DAT is StartID and SkipID. In professional mode, there is an SDLC protocol or, if DAT, it may be the same as consumer mode." The physical connection media are commonly used with AES/EBU: balanced (differential), using two wires and shield in three-wire microphone cable with XLR connectors; unbalanced (single-ended), using audio coax cable with RCA jacks; and optical (via fiber optics). S/P-DIF "S/P-DIF" (Sony/Philips Digital Interface Format) typically refers to AES/EBU operated in consumer mode over unbalanced RCA cable. Note that S/P-DIF and AES/EBU mean different things depending on how much of a purist you are in the digital audio world; see the Finger article below. References: Finger, Robert, AES3-199X: The Revised Two Channel Digital Audio Interface (DRAFT), presented at the 91st Convention of the Audio Engineering Society, October 4-8, 1991. Reprints: AES, 60 East 42nd St., New York, NY, 10165. [The above from Phil Lapsley and Tim Channon, tchannon@black.demon.co.uk] Painter, E. M., and Spanias, A. S. (1997 and revised 1999). A Review of Algorithms for Perceptual Coding of Digital Audio Signals. (PostScript, 3MB) http://www.eas.asu.edu/~spanias/papers.html [Andreas Spanias, spanias@asu.edu] ---------------------------------------------------------------------- Q2.7: What is mu-law encoding? Where can I get source for it? Updated 9/13/99 Mu-law (also "u-law") encoding is a form of logarithmic quantization or companding. It's based on the observation that many signals are statistically more likely to be near a low signal level than a high signal level. Therefore, it makes more sense to have more quantization points near a low level than a high level. In a typical mu-law system, linear samples of 14 to 16 bits are companded to 8 bits. Most telephone quality codecs (including the Sparcstation's audio codec) use mu-law encoded samples. Desktop Sparc machines come with routines to convert between linear and mu-law samples. On a desktop Sparc, see the man page for audio_ulaw2linear in /usr/demo/SOUND/man. To obtain: Craig Reese posted the source of similar routines to comp.dsp in August '92. These are archived on ftp://mirriwinni.cse.rmit.edu.au/pub/dsp/misc/ulaw_reese. References: ITU-T (formerly CCITT) Recommendation G.711 (very difficult to follow). Michael Villeret, et. al, A New Digital Technique for Implementation of Any Continuous PCM Companding Law, IEEE Int. Conf. on Communications, 1973, vol. 1, pp. 11.12-11.17. MIL-STD-188-113, Interoperability and Performance Standards for Analog-to-Digital Conversion Techniques, 17 February 1987. TI Digital Signal Processing Applications with the TMS320 Family (TI literature number SPRA012A), pp. 169-198. [From Joe Campbell; Craig Reese, cfreese@super.org; Sepehr Mehrabanzad, sepehr@falstaff.dev.cdx.mot.com; Keith Kendall, KLK3%mimi@magic.itg.ti.com] ---------------------------------------------------------------------- Q2.8: How can I do CD <=> DAT sample rate conversion? Updated 9/13/99 CD players use a 44.1 kHz sample rate, whereas DAT uses a 48 kHz sample rate. This means that you must do sample rate conversion before you can get data from a CD player directly into a DAT deck. [From Ed Hall, edhall@rand.org:] For a start, look at Multirate Digital Signal Processing by Crochiere and Rabiner (see FAQ section 1.1). Almost any technique for producing good digital low-pass filters will be adaptable to sample-rate conversion. 44.1:48 and vice-versa is pretty hairy, though, because the lowest whole-number ratio is 147:160. To do all that in one go would require a FIR with thousands of coefficients, of which only 1/147th or 1/160th are used for each sample--the real problem is memory, not CPU for most DSP chips. You could chain several interpolators and decimators, as suggested by factoring the ratio into 3*7*7:2*2*2*2*2*5. This adds complexity, but reduces the number of coefficients required by a considerable amount. [From Lou Scheffer:] Theory of operation: 44.1 and 48 are in the ratio 147/160. To convert from 44.1 to 48, for example, we (conceptually): 1. interpolate 159 zeros between every input sample. This raises that data rate to 7.056 MHz. Since it is equivalent to reconstructing with delta functions, it also creates images of frequency f at 44.1-f, 44.1+f, 88.2-f, 88.2+f, ... 2. We remove these with an FIR digital filter, leaving a signal containing only 0-20 KHz information, but still sampled at a rate of 7.056 MHz. 3. We discard 146 of every 147 output samples. It does not hurt to do so since we have no content above 24 KHz. In practice, of course, we never compute the values of the samples we will throw out. So we need to design an FIR filter that is flat to 20 KHz, and down at least X db at 24 KHz. How big does X need to be? You might think about 100 db, since the max signal size is roughly +-32767, and the input quantization +- 1/2, so we know the input had a signal to broadband noise ratio of 98 db at most. However, the noise in the stopband (20KHz-3.5MHz) is all folded into the passband by the decimation in step 3, so we need another 22 db (that's 160 in db) to account for the noise folding. Thus 120 db rejection yields a broadband noise equal to the original quantizing noise. If you are a fanatic, you can shoot for 130 db to make the original quantizing errors dominate, and a 22.05 KHz cutoff to eliminate even ultrasonic aliasing. You will pay for your fanaticism with a penance of more taps, however. To obtain: There's a free implementation of Julius O. Smith III and someone else's "bandwidth-limited interpolation" rate conversion algorithm. A paper available as ftp://ccrma-ftp.stanford.edu/pub/DSP/Tutorials/BandlimitedInterpolation.eps.Z explains the algorithm. Free source code, as well as an HTML discussion of the algorithm, is available at http://ccrma-www.stanford.edu/~jos/resample/. It all works quite well. [From Kevin Bradley, kb+@andrew.cmu.edu:] There is an implementation of polyphase resampling for various rates as a part of the Sox audio toolkit at http://home.sprynet.com/~cbagwell/sox.html. See file polyphas.c for details. Sox also contains an implementation of bandlimited interpolation and linear interpolation, and serves as a ready vehicle for module experimentation. [From Fritz M. Rothacher, f.rothacher@ieee.org:] You can add my Ph.D. thesis on sample-rate conversion to the FAQ: Fritz M. Rothacher, Sample-Rate Conversion: Algorithms and VLSI Implementation, Ph.D. thesis, Integrated Systems Lab, Swiss Federal Institute of Technology, ETH Zuerich, 1995, ISBN 3-89191-873-9 It can also be downloaded from my homepage at http://www.guest.iis.ee.ethz.ch/~rota. ---------------------------------------------------------------------- Q2.9: Wavelets Updated 6/3/98 Q2.9.1 What are wavelets? Where can I get more information? In short, wavelets are a way to analyze a signal using base functions which are localized both in time (as diracs, but unlike sine waves), and in frequency (as sine waves, but unlike diracs). They can be used for efficient numerical algorithms and many DSP or compression applications. Sources of information on wavelets include: * a newsletter, "Wavelet Digest". Subscriptions for Wavelet Digest: E-mail to wavelet@math.scarolina.edu with "subscribe" as subject. The Wavelet Digest can also be found at http://www.wavelet.org/. * http://www.amara.com/current/wavelet.html ---------------------------------------------------------------------- Q2.9.2 What are some good books and papers on wavelets The best introduction to wavelet transforms is in: Wavelets and Signal Processing- Oliver Rioul and Martin Vetterli, IEEE Signal Processing magazine, Oct. 91, pp 14-38 A good introductory book on wavelets: Randy K. Young, Wavelet Theory and Its Applications, Kluwer Academic Publishers, ISBN 0-7923-9271-X, 1993. A more thorough book: Ali N. Akansu and Richard A. Haddad, Multiresolution Signal Decomposition Transforms, Subbands, Wavelets Academic Press, Inc., ISBN 0-12-047140-X A couple more interesting papers: Wavelets and Filter banks: Theory and Design, IEEE Transactions on Signal Processing, Vol. 40, No.9, Sept. 1992, pp 2207-2232 Mac Cody's articles in Dr. Dobb's Journal, April 1992 and April 1993 Paper by Ingrid Daubechies in IEEE Trans. on Info. theory , vol 36. No.5 , Sept 1990 and a book titled " Ten lectures on Wavelets" deal with the mathematical aspects of the WT. ---------------------------------------------------------------------- Q2.9.3: Where can I get some software for wavelets? ftp://pascal.math.yale.edu/pub/wavelets/software/xwpl Binaries are available for the following platforms: Sun Sparcstations running SunOS 4.1 or Solaris 2.3, NeXT machines running NeXTstep 3.0 or higher, with an X server, Silicon Graphics machines (IRIS), DEC Alpha AXP running OSF/1 1.2 or higher, i386/i486 PC compatible with Linux 0.99. There is also a sample data directory containing interesting signals. More information: http://www.math.yale.edu/users/majid/ [From Fazal Majid majid@math.yale.edu]: Rice Wavelet Tools Description: The Rice Wavelet Toolbox (RWT) is a collection of Matlab M-files and C MEX-files for 1D and 2D wavelet and filter bank design, analysis, and processing. The toolbox provides tools for denoising and interfaces directly with our Matlab code for wavelet domain hidden Markov models and wavelet regularized deconvolution. Also included is a simple converter to the data format used by the official Matlab wavelet toolbox. The current distribution, Version 2.3 (Dec 1, 2000), has been streamlined and packaged for different systems, including Solaris, Linux, and Microsoft Windows. Functions omitted in Version 2.3 can be found in the Version 2.01 distribution. To obtain: See http://www-dsp.rice.edu/software/RWT/. Send mail to wlet-tools@rice.edu (or ramesh@dsp.rice.edu) ---------------------------------------------------------------------- Q2.10: How do I calculate the coefficients for a Hilbert transformer? Updated 6/3/98 For all the gory details, I suggest the paper: Andrew Reilly and Gordon Frazer and Boualem Boashash: Analytic signal generation---tips and traps, IEEE Transactions on Signal Processing, no. 11, vol. 42, Nov. 1994, pp. 3241-3245. For comp.dsp, the gist is: 1. Design a half-bandwidth real low-pass FIR filter using whatever optimal method you choose, with the principle design criterion being minimization of the maximum attenuation in the band f_s/4 to f_s/2. 2. Modulate this by exp(2 pi f_s/4 t), so that now your stop-band is the negative frequencies, the pass-band is the positive frequencies, and the roll-off at each end does not extend into the negative frequency band. 3. either use it as a complex FIR filter, or a pair of I/Q real filters in whatever FIR implementation you have available. If your original filter design produced an impulse response with an even number of taps, then the filtering in 3 will introduce a spurious half-sample delay (resampling the real signal component), but that does not matter for many applications, and such filters have other features to recommend them. Andrew Reilly [Reilly@zeta.org.au] ---------------------------------------------------------------------- Q2.11: Algorithm implementation: floating-point versus fixed-point According to the WWWebster Dictionary, an algorithm is "a procedure for solving a mathematical problem (as of finding the greatest common divisor) in a finite number of steps that frequently involves repetition of an operation; broadly: a step-by-step procedure for solving a problem or accomplishing some end especially by a computer." Typical (although by no means the only) operations are those of addition and multiplication. When expressing the algorithm with pencil and paper, these operations are commonly taken to be within an algebraically complete number system such as the integers or the reals. However, when the time comes to implement the algorithm on a computer, these "ideal" number systems must be exchanged for something realizable. The number systems available today on common processors and digital hardware are broadly categorized as floating-point and fixed-point. In a floating-point representation, the total number of bits available are partitioned into an exponent and mantissa. Generally speaking, the mantissa stores the "significant digits" of the value while the exponent scales the significant digits to the desired magnitude. The action of the exponent is to move, or "float," the decimal point depending on the magnitude being represented; thus the term "floating-point." Because floating-point representations are typically at least 32 bits long (IEEE-754 is a popular standard for 32-bit and 64-bit floating-point numbers), there exists simultaneously high precision and high dynamic range. These traits of floating-point numbers allow most algorithms to be ported directly to floating-point implementations with little or no change, and this is the key reason floating-point representations are highly desirable. The disadvantage of floating-point implementations is that they require a significant amount of extra hardware over fixed-point implementations, which translates to higher parts costs, higher power consumption, slower execution, larger chip area, or a combination of these. As the term "fixed-point" implies, fixed-point representations have the binary point at a fixed location. There are two subsets of fixed-point implementations: fractional and integer. In a fractional fixed-point implementation, such as that provided on the Motorola 56K series of DSPs, the binary point is always assumed to be to the left of the most-significant digit. In an integer fixed-point implementation, such as that provided by the Texas Instruments TMS320C54xx series of DSPs, the binary point is to the right of the least-significant digit. In either case, the arithmetic operations implemented in the hardware are essentially integer, which results in a much simpler arithmetic logic unit in hardware that allows lower cost, lower power consumption, faster execution, smaller chip area, or a combination of these, over that of floating-point implementations. For more information on the IEEE-754 format, see James Carr's "Numerical Analysis" class notes at Florida State University, http://www.scri.fsu.edu/~jac/MAD3401/Backgrnd/ieee.html. Fixed-Point Arithmetic: The Basics In essence, a fixed-point representation is a simple integer scaled (divided) by a power of two. If we denote an unscaled integer variable by upper case "X" and the scaled, fixed-point variable by lower case "x," then x = X/2^b, where b is the number of digits the binary point is shifted left. For example, if X is a 16-bit, two's complement integer, and b=4, then "X" has values ranging from -2^(15) to +2^(15)-1 and with minimum step size of 1, while the scaled value "x" ranges from -2^(11) to +2^(11) - 1/(2^4) with a minimum step size of 1/(2^4). Note that the value of "b" is not part of the representation. You won't see it in a register or as part of the data anywhere; it is a parameter that the algorithm implementer must determine and maintain. Fixed-point representations place some very different rules on operations than their floating-point counterparts. For example, two variables must be scaled the same in order to be added (or subtracted). Thus it may be necessary to shift one or the other operand prior to adding. Another example is that when multiplying two N-bit values with scale factors b0 and b1, the result is scaled (b0+b1) and requires 2*N bits in general in order to avoid overflow and maintain precision. There are several other rules and considerations for fixed-point arithmetic that are commonly encountered when implementing algorithms. For more information, see http://home.earthlink.net/~yatescr/papers.htm. Randy Yates [yates@ieee.org] Previous section (1) Next section (3) Previous section (2) Next section (4) Q3: Programmable DSP chips and their software Q3.1: What are the available DSP chips and chip architectures? Updated 05/07/02 The "big four" programmable DSP chip manufacturers are Texas Instruments, with the TMS320C2000, TMS320C5000, and TMS320C6000 series of chips; Motorola, with the DSP56300, DSP56800, and MSC8100 (StarCore) series; Agere Systems (formerly Lucent Technologies), with the DSP16000 series; and Analog Devices, with the ADSP-2100 and ADSP-21000 ("SHARC") series. A good overview of programmable DSP chips is published periodically in EDN and Computer Design magazines. You may also want to check out Berkeley Design Technology's home page, which has a number of articles on choosing DSP processors, as well as a "Pocket Guide to DSP Processors and Cores" in both HTML and PDF formats. Brief overviews of various DSP processors, cores, and general-purpose processors can be found at http://www.bdti.com/procsum/index.htm. Here's a less ambitious chip breakdown by manufacturer: Agere Systems (formerly Lucent Technologies): DSP16xxx: 100 to 170 MHz 16-bit fixed-point DSP. The DSP16000 core features two multipliers with SIMD-like capabilities, a 20-bit address bus, a 32-bit address bus, and eight 40-bit accumulators. The chips feature two serial ports and two timers. The first-generation processor, the DSP16210, contains a single DSP16000 core and 120 KB of internal RAM. The second-generation DSP16410 incorporates two DSP16000 cores and 386 KB of internal RAM. Analog Devices: ADSP-21xx: 10 to 80 MHz 16-bit fixed point DSPs; 40-bit accumulator; 24-bit instructions. Large number of family members with different configurations of on-chip memory and serial ports, timers, and host ports. ADSP-21mspxx members include an on-chip codec. ADSP-219x: 160 MHz 16-bit fixed point DSPs; 40-bit accumulator; 24-bit instructions. Based on the ADSP-21xx family, and is is mostly, but not completely, assembly source-code upward compatible with the ADSP-21xx Adds new addressing modes and an instruction cache, expands address space, and lengthens pipeline (six stages compared to three on the ADSP21xx). Family includes members containing multiple ADSP-219x cores. ADSP-21xxx ("SHARC"): 33 to 100 MHz floating-point DSP; Supports 32-bit fixed-point, IEEE format 32-bit floating-point, and 40-bit floating-point; 40-bit registers plus an 80-bit accumulator that can be divided into two 32-bit registers and a 16-bit register. The first-generation SHARC, the ADSP-2106x, features a single data path, a 32-bit address bus, and 40-bit data bus. Versions are available with up to 512 KB of on-chip memory, up to six communication ports, and up to 10 DMA channels. The second-generation ADSP-2116x has two parallel data paths, a 32-bit address bus, and a 64-bit data bus. Versions are available with up to 512 KB of on-chip memory; up to six communication ports, and up to 14 DMA channels. Analog Devices also sells the AD14000 series, which contain four ADSP-2106x SHARC processors in a single-chip package. ADSP-2153x: 200 to 300 MHz 16-bit fixed point DSPs that can execute two MAC instructions per cycle; based on the ADI/Intel MSA core. Uses a mix of 16-, 32-, and 64-bit instructions. Features include ability to operate over a wide range of frequencies and voltages. Motorola: DSP563xx: 66 to 160 MHz 24-bit fixed-point DSP; most family members have 24-bit address and data busses. The DSP563xx also features 56-bit accumulators (2), timers, serial interface, host interface port. The DSP56307 and DSP56311 contain a filter co-processor. Up to 1 MB of internal RAM. DSP568xx: 40 MHz 16-bit fixed point DSP; 36-bit accumulators (2), three internal address buses (two 16-bit, one 19-bit) and one 16-bit external address bus; three 16-bit internal data buses, one 16-bit external data bus; serial ports, timers. 4-12 KB of internal RAM. Most family members include an on-chip A/D. DSP5685x: 160 MHz 16-bit fixed point DSP based on the DSP568xx. Adds an exponent detector and two accumulators, extends shifter and the logic unit to 32 bits, and widens internal address and data buses. The DSP5685x uses a 1X master clock rate rather than the 2X master clock rate used by the DSP568xx. MSC81xx: The 300 MHz MSC8101 is the first processor based on the StarCore SC140 core. It contains four parallel ALU units that can execute up to four MAC operations in a single clock cycle. The MSC8101 uses variable-length instructions. Features include: 512 KB on-chip RAM; 16 DMA channels; an on-chip filter co-processor; and interfaces for ATM, Ethernet, E1/T1 and E3/T3, and the PowerPC bus. Texas Instruments: TMS320C2xxx: 20-40 MHz 16-bit fixed-point DSPs oriented toward low-cost control applications; 16 bit data, 32 bit registers. The family members have a variety of peripherals, such as A/D converters, 41 I/O pins, and 16 PWM outputs. A variety of RAM and ROM configurations are available TI also sells the TMS320C2x family, an older version of the chip with fewer features. TMS320C3x: 33-75 MHz floating point DSPs; 32-bit floating-point, 24-bit fixed-point data, 40-bit registers; DMA controller; serial ports; some support for multi-processor arrays. Various ROM and RAM configurations. TMS320C54xx: 40 to 160 MHz 16-bit fixed-point DSPs with a large number of specialized instructions. Many family members; the processors differ in configuration of on-chip ROM/RAM, serial ports, autobuffered serial ports, host ports, and time-division multiplexed ports. On-chip RAM ranges from 10 KB to over 1 MB. TMS320C55xx: 144 to 200 MHz dual-ALU variant of the TMS320C54xx that can execute two MAC instructions per cycle. Variable instruction word width. Features include up to 320 KB internal RAM; 6 DMA channels; 2 serial ports; and 2 timers. TMS320C62xx: 150-300 MHz 16-bit fixed-point DSP with VLIW (very large instruction word), load/store architecture; 32 32-bit registers; very deep pipeline; two multipliers, ALUs, and shifters; cache. TMS320C64xx: 400-600 MHz 16-bit fixed-point DSP based on the TMS320C62xx. Adds SIMD support to most execution units, including extensive 8-bit SIMD support. Also doubles data bandwidth and increases size of on-chip memory. TMS320C67xx: 100-167 MHz 32-bit and 64-bit IEEE-754 floating-point DSP with VLIW (very large instruction word), load/store architecture; 32 32-bit registers; very deep pipeline; two multipliers, ALUs, and shifters; cache. ---------------------------------------------------------------------- Q3.2: What is the difference between a DSP and a microprocessor? Updated 04/02/01 The essential difference between a DSP and a microprocessor is that a DSP processor has features designed to support high-performance, repetitive, numerically intensive tasks. In contrast, general-purpose processors or microcontrollers (GPPs/MCUs for short) are either not specialized for a specific kind of applications (in the case of general-purpose processors), or they are designed for control-oriented applications (in the case of microcontrollers). Features that accelerate performance in DSP applications include: * Single-cycle multiply-accumulate capability; high-performance DSPs often have two multipliers that enable two multiply-accumulate operations per instruction cycle; some DSP have four or more multipliers * Specialized addressing modes, for example, pre- and post-modification of address pointers, circular addressing, and bit-reversed addressing * Most DSPs provide various configurations of on-chip memory and peripherals tailored for DSP applications. DSPs generally feature multiple-access memory architectures that enable DSPs to complete several accesses to memory in a single instruction cycle * Specialized execution control. Usually, DSP processors provide a loop instruction that allows tight loops to be repeated without spending any instruction cycles for updating and testing the loop counter or for jumping back to the top of the loop * DSP processors are known for their irregular instruction sets, which generally allow several operations to be encoded in a single instruction. For example, a processor that uses 32-bit instructions may encode two additions, two multiplications, and four 16-bit data moves into a single instruction. In general, DSP processor instruction sets allow a data move to be performed in parallel with an arithmetic operation. GPPs/MCUs, in contrast, usually specify a single operation per instruction While the above differences traditionally distinguish DSPs from GPPs/MCUs, in practice it is not important what kind of processor you choose. What is really important is to choose the processor that is best suited for your application; if a GPP/MCU is better suited for your DSP application than a DSP processor, the processor of choice is the GPP/MCU. It is also worth noting that the difference between DSPs and GPPs/MCUs is fading: many GPPs/MCUs now include DSP features, and DSPs are increasingly adding microcontroller features. [Ole Wolf, wolf@bdti.com] ---------------------------------------------------------------------- Q3.3: Software for Analog Devices DSPs Updated 05/08/02 Q3.3.1: Where can I get a C compiler for the ADSP-21xx and ADSP-21xxx? The G21 package collects the free source code for the Analog Devices GCC-based C compilers for their 21xxx (SHARC) and 21xx series DSPs. These compilers are all based on GCC version 2.3.3. Full source code for the compiler, assembler, linker, etc. is available at http://www.kvaleberg.com/g21.html. The C compilers are available for the 210x series as well as for the SHARC. The assemblers and linkers are only available for the SHARC. The source code is based on what is released under GPL by ADI, but is adapted for use with Linux and other Unix variants. [Egil Kvaleberg, egil@kvaleberg.no] ---------------------------------------------------------------------- Q3.3.2: Where can I get tools for the ADSP-21xxx? SHARC development tools are avaiable for Acorn/BSD, Linux, and other platforms. The tools include frontend/preprocessor , assembler, linker, archiver, a utility to generate ROM images for eprom burners, and other utilities The supplied assembler is not part of the gnu archive, but is based on a assembler originaly written by P. Lantto. Source code and binaries are available at http://www.ww.tu-freiberg.de/~pberg/fgm/support/index.html. [Theo Markettos atm26@cam.ac.uk] ---------------------------------------------------------------------- Q3.3.3: Where can I get an assembler for the ADSP-2105? John Sture has developed an assembler for the Analog Devices ADSP-2105. The latest version can be obtained from http://www.suresoft.ca. (Follow the links to the FTP site and select beta0.4.1src.tar.gz.) Requires Analog Devices' ld21.exe version 5.1 linker or equivalent for linking executables. Source code to the assembler is included under the terms of the GNU public license. [John Sture, jsture@vcn.bc.ca] ---------------------------------------------------------------------- Q3.3.4: Where can I get algorithms or libraries for Analog Devices DSPs? The number for the Analog Devices DSP BBS is (617) 461-4258 (300, 1200, 2400, 9600, 14400 bps), 8N1. You can also find files on Analog Devices' web site at http://www.analog.com/support/Design_Support.html, or at their FTP site at ftp://ftp.analog.com. [Analog Devices DSP Applications, dsp_applications@analog.com] ---------------------------------------------------------------------- Q3.4: Software for Agere Systems (Formerly Lucent Technologies) DSPs Agere Systems provides application libraries for their DSPs at http://www.lucent.com/micro/wam/tse/. ---------------------------------------------------------------------- Q3.5: Software for Motorola DSPs Updated 05/07/02 Motorola provides free software development tools that may be downloaded from the Motorola Web site at http://e-www.motorola.com/webapp/sps/site/prod_summary.jsp?code=MSW3SDK000AA&nodeId=01M983916044937. Q3.5.1: Where can I get a free assembler for the Motorola DSP56000? A free assembler for the Motorola DSP56000 exists, thanks to Quinn Jensen, jensenq@zdomain.com. The current version is 1.2. It is also available at ftp://ftp.zdomain.com/pub/jensenq/a56 and http://www.zdomain.com/a56.html. ---------------------------------------------------------------------- Q3.5.2: Where can I get a free C compiler for the Motorola DSP56000? There are two separate compiler sources for the Motorola DSP56000. One is the port of gcc 1.40 done by Andrew Sterian (asterian@umich.edu) and the other is a port of gcc 1.37.1 done by Motorola and returned to the FSF. Andrew's port has bowed to Motorola's version. Both may be portable to gcc2.x.x with some effort required. Neither of these comes with an assembler, but you can get a free DSP56000 assembler elsewhere (see question 3.5.1, above). The Motorola gcc source is available for FTP from: ftp://nic.funet.fi/pub/ham/dsp/dsp56k-tools/dsp56k-gcc.tar.Z and ftp://mirriwinni.cse.rmit.edu.au/pub/uP/56k/g56k.tar.gz-1996. From Andrew Sterian, asterian@umich.edu: "My DSP56K compiler, while not supported nor as well tested as Motorola's, implements fixed-point arithmetic rather than floating-point arithmetic. This may be suitable for some applications. The 5615 compiler also implements fixed-point arithmetic. To the best of my knowledge, Motorola does not have a C compiler for the 5615 family, although alternatives may exist. As of this writing (January 1997) I have not worked with Motorola DSPs or compiler software for nearly 5 years so questions regarding my compilers may well be met with "Ummm... I have no idea." Both compilers were posted to alt.sources so any Usenet site that archives this newsgroup will have a copy. I have also found the 5616 compiler at ftp://ftp.funet.fi/pub/ham/dsp/dsp56k-tools/gcc5616.tar.Z. href="http://www.newmicros.com">http://www.newmicros.com) IsoPod(TM) - based on the DSP56F805. The assembler generates output suitable for Motorola's free JTAG flash loader. Pete Gray has announced the availability of a Small C cross-compiler (with source) and assembler for the Motorola DSP56800, available for download from http://home.attbi.com/~petegray. Targetting a simple DOS-box host, developed and tested using djgpp (http://www.delorie.com/djgpp/) and Metrowerks CodeWarrior, in conjunction with NMI's (http://www.newmicros.com) IsoPod(TM) - based on the DSP56F805. The assembler generates output suitable for Motorola's free JTAG flash loader. Small C language reference available online at http://www.ddjembedded.com/languages/smallc/ Pete also asks for comments and feedback to be sent to petegray@ieee.org ---------------------------------------------------------------------- Q3.5.3 Where can I get a disassembler for the Motorola DSP56000? Miloslaw Smyk has released an open source (BSD style) 5600x disassembly library. It is available for download at https://sourceforge.net/projects/lib5600x [Miloslaw Smyk, thorgal@wmfh.org.pl] ---------------------------------------------------------------------- Q3.5.4: Where can I get algorithms and libraries for Motorola DSPs? Motorola provides a software archive that is available via World-Wide Web from the software page at http://www.mot.com/SPS/DSP/software/. The archive includes macros for filters (FIR, IIR, adaptive) and floating-point functions. [Tim Baggett] ---------------------------------------------------------------------- Q3.5.5: Where can I get NeXT-compatible Motorola DSP56001 code? Try FTP at ccrma-ftp.stanford.edu. The /pub/ directory contains free code for the Motorola DSP56001 and the NeXT platform. [bil@ccrma.Stanford.EDU] ---------------------------------------------------------------------- Q3.5.6: Where can I get emulators for the 68HC11 (6811) processor? While the 68HC11 is not a DSP processor, emulators are available for those who might be interested in doing DSP on these processors: * New Mexico State University (NMSU) simulator engine, ftp://crl.nmsu.edu/pub/non-lexical/6811/ (Unix). Simulator engine with a command-line interface. * Sim6811, ftp://cherupakha.media.mit.edu/pub/projects/6811/sim6811/ (Mac). Screen-oriented user interface based on the NMSU simulator engine (plus bug fixes). * THRSim11, http://programfiles.com/index.asp?ID=8366 allows you to edit, assemble, simulate and debug programs for the 68HC11 on Windows 95/98. THRSim11 simulates the CPU, ROM, RAM, all memory mapped I/O ports, and the on board peripherals. ---------------------------------------------------------------------- Q3.6: Software for Texas Instruments DSPs Updated 05/07/02 Q3.6.1: Where can I get free algorithms or libraries for TI DSPs? nic.funet.fi has some old, apparently public domain, assembler and related tools from TI for the TMS320 family. The directory is /pub/ham/dsp. [Antti-Pekka Virtanen, antsu@utu.fi] TI has a number of free algorithms available on their website at http://dspvillage.ti.com/docs/sdstools/sdscommon/showsdsinfo.jhtml?templateId=57&path=templatedata/cm/ccstudio/data/free_tools. TI's world-wide web site is http://www.ti.com. The TI DSP bulletin board is mirrored on ftp.ti.com, and on mirriwinni.cse.rmit.edu.au. The TI site is the official one, but has no user contributed software. The file: ftp://mirriwinni.cse.rmit.edu.au/mirrors/tibbs/00readme (might be broken) provides further guidance. Please restrict FTP session to outside of 8 am to 6 pm local time (10 pm to 8 am GMT). [Brad Hards, bradh@gil.com.au] { If anyone knows of any other sources for TI DSP software, please let us know at comp-dsp-faq@bdti.com. Thanks! } ---------------------------------------------------------------------- Q3.6.2: Where can I get free development tools for TI DSPs? TI development tools are available for free 30 day evaluation on the TI website. Go to http://www.ti.com/sc/docs/tools/index.htm. ---------------------------------------------------------------------- Q3.6.3: Where can I get a free C compiler for the TI TMS320C3x/4x? The GNU binutils 2.11 and later have been ported to the TI C54xx/IBM C54DSP. Most of the binutils tools are supported, including the assembler, linker and objdump. The assembler is source-compatible with the TI assembler. The GNU binutils are available from http://sources.redhat.com/binutils/ GDB ports for c25/c5x/c54x are also available. [Timothy Wall] Dr. Michael P. Hayes has written a GNU C-based compiler for the TMS320C30 and TMS320C40 families, available at http://www.elec.canterbury.ac.nz/c4x. The current version patches against gcc-2.8.1; support is moving to egcs-1.2. The compiler is freely redistributable under the terms of the GNU Public License. Front-ends are also available for C++, Java, Fortran 77, Pascal, Ada 95, among others. [Dr. Michael P. Hayes, m.hayes@elec.canterbury.ac.nz] ---------------------------------------------------------------------- Q3.6.4: Where can I get a free assembler for the TI TMS320C3x/4x? Ted Rossin has written an assembler and linker for the TMS320C30. In his words, "It is somewhat limited by the fact that it can't handle expressions but it has worked fine for me over the past few years. There is no manual because it is a clone of the TI assembler and linker. However the linker command files use a different (easier to use) syntax. It runs on HP-UX workstations, Macs, IBM clones and believe it or not the Atari-ST (because I developed the code on it)." [Ted Rossin, rossin@fc.hp.com] Dr. Michael P. Hayes has written a GNU-based assembler for the TMS320C30 and TMS320C40 families, available at http://www.elec.canterbury.ac.nz/c4x. The current version patches against binutils-2.7. According to Michael Hayes, the assembler syntax is compatible with the Texas Instruments TMS320C30 assembler, although not all the Texas Instruments directives are supported. The binutils include a linker (ld), archiver (ar), disassembler (objdump), and other miscellaneous utilities. The object format of the assembler is compatible with the COFF format used by the Texas Instruments assembler. The assembler and other binary utilities are freely redistributable under the terms of the GNU Public License. [Dr. Michael P. Hayes, m.hayes@elec.canterbury.ac.nz] ---------------------------------------------------------------------- Q3.6.5: Where can I get a free simulator for the TI TMS320C3x/4x? A freely distributable instruction set architecture simulator is available for the TMS320C30 DSP as part of the Web-Enabled Simulation framework from UT Austin at http://anchovy.ece.utexas.edu/~arifler/wetics/. We have released all of the source code, as well as prebuilt C30 simulators for Windows '95/NT and Solaris 2.5 architectures. The C30 simulator is bit-, cycle-, and instruction-accurate. The behavior of the C30 simulator has been validated against a C30 DSK board. The C30 simulator correctly reports interlocking and pipeline flushes, so it provides a convenient way to check C30 programs for these hidden delays. The C30 simulator is based on the C30 DSK tools by Keith Larson at Texas Instruments. [Brian Evans, bevans@combo.ece.utexas.edu] Herman Ten Brugge (haj.ten.brugge@net.hcc.nl) has also written a GNU debugger (GDB) based simulator for the TMS320C30 and TMS320C40, available via anonymous FTP at http://www.elec.canterbury.ac.nz/c4x. This is freely redistributable under the terms of the GNU Public License. This simulator allows you to debug your programs without having to a connect to a real C[34]x target system. It will also profile your code showing you where the pipeline conflicts are occurring. You can connect I/O ports to files (or TCP/IP sockets), trigger interrupts, examine the cache etc. It will detect different threads of control running and generate a profile summary for each thread, annotating both the C code and assembler code with the number of executed cycles. [Dr. Michael P. Hayes, m.hayes@elec.canterbury.ac.nz] ---------------------------------------------------------------------- Q3.6.6: What is Tick? Where can I get it? Tick is a TMS320C40 parallel network detection and loader utility. It is available from: ftp://mirriwinni.cse.rmit.edu.au/mirrors/tibbs/UserContributed. Supports: Transtech, Hunt, and Traquair boards hosted by DOS, SunOS, Linux (a PC unix) Previous section (2) Next section (4)