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comp.dsp FAQ [1 of 4]

Started by Unknown May 4, 2006
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,
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)
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                 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)

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