Internet Info 1993/Internet Info CD-ROM (Walnut Creek) (1993).iso/answers/comp/dsp-faq/part1

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Subject: comp.dsp FAQ [1 of 4]
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Archive-name: dsp-faq/part1
Last-modified: Sun May 30 1993
Version: 1.1

Welcome to the comp.dsp FAQ.  We are seeking information on all topics,
from the best layout (e.g., should chips and prototype boards be
another FAQ?) to a good annotated list of DSP articles and reference
books.  Please email your suggestions to [e-mail].
Meta-comments are all in curly braces, {like this}.  Acknowledgements to
people responsible for articles and/or bits of information are in
square brackets, [like so].

You can ftp the very latest version of this FAQ from: evans.ee.adfa.oz.au
(131.236.30.24) in pub/dsp/dsp-faq.help, or from copernicus.Berkeley.EDU
in dsp-faq.help.*.

In version 1.1 of the FAQ, we've included a list of DSP-related FTP sites
at the end of the FAQ (section 7).  Be sure to check this if you're looking
for an FTP site, as it may have more up-to-date information on than the text
in the body the FAQ.

We are most grateful to the following contributors who helped out with
this version of the FAQ:

Harry Bates        [e-mail]
Chuck Corley        [e-mail]
Jutta Degener        [e-mail]
John Edwards        [e-mail]
Cole Erskine        [e-mail]
Robert Gammon        [e-mail]
Alan Kamas        [e-mail]
Stephen Moshier        [e-mail]
Ted Rossin        [e-mail]
Paul Rubin        [e-mail]
Witold Waldman        [e-mail]

and the following folks who helped out with previous versions:

William Alves        [e-mail]
Harry Bates        transtech![e-mail]
Annard Brouwer        [e-mail]
Joe Campbell        [e-mail]
Tim Channon        [e-mail]
Steve Clift        [e-mail]
Joel Coltoff        [e-mail]
Chuck Corley        [e-mail]
Jutta Degener        [e-mail]
Malachy Devlin        [e-mail]
John Edwards            [e-mail]
Kenneth Ekman        [e-mail]
Brian Evans        [e-mail]
John Fisher        [e-mail]
Dan Frankowski        [e-mail]
Robert Gammon        [e-mail]
Kevin Gardner        [e-mail]
Maurice Givens        [e-mail]
Ed Hall            [e-mail]
Vince Herried        [e-mail]
Eric Jacobsen        [e-mail]
Pete Janzow        [e-mail]
Quinn Jensen        [e-mail]
Richard Kim        [e-mail]
Gints Klimanis        [e-mail]
Greg Koker        [e-mail]
Juhana Kouhia        [e-mail]
John Lazzaro        [e-mail]
Sergio Liberman        [e-mail]
Stan McClellan          [e-mail]
Renzo Mercanzin        [e-mail]
Phillip Musumeci    [e-mail]
Bill Ralston        [e-mail]
C.S. Ramalingam        [e-mail]
Tony Richardson        [e-mail]
Ted Rossin        [e-mail]
Paul Russell        [e-mail]
Stanley Sasaki        [e-mail]
Bill Schottstaedt    [e-mail]
Paul Simoneau        [e-mail]
Greg Smart        [e-mail]
Larry Stewart        [e-mail]
Andrew Ukrainec        [e-mail]
Witold Waldman        [e-mail]
Phil Woodland        [e-mail]
Ronnin Yee        [e-mail]

Get your name here - contribute something!  (If you don't want your
email address listed in your contribution, please indicate so.)

Phil Lapsley       [e-mail]  (FAQ maintainer)
Bradley Hards       [e-mail]   (assistant/original FAQ maintainer)

=============================================================================

0. What is comp.dsp?

1. General DSP.
   1.1 DSP book and article references.
       1.1.1 Bibles of DSP.
       1.1.2 Adaptive signal processing.
       1.1.3 Array signal processing.
       1.1.4 Windowing.
       1.1.5 Digital audio effects processing.
   1.2 Where can I get free software for general DSP?
       1.2.1 What is Gabriel?  Where can I get it?
       1.2.2 What is Ptolemy?  Where can I get it?
       1.2.3 What is Khoros?  Where can I get it?
       1.2.4 What are DSP Tutorials?  Where can I get them?
       1.2.5 What are some DSP extensions to MATLAB?  Where can I get them?
       1.2.6 What is the Signal Processing Package for Mathematica?
             Where can I get it?
       1.2.7 What is the Controls System Package for Mathematica?
             Where can I get it?
       1.2.8 What are some other DSP Notebooks for Mathematica?
       1.2.9 What is the Linear Systems Toolbox for Maple?  Where can I get it?
       1.2.10 Where can I get text to speech conversion software?
       1.2.11 Where can I get filter design software?
       1.2.12 What is SigLib? Where can I get it?
       1.2.13 What is the AudioFile System?  Where can I get it?

2. Algorithms and standards.
   2.1 Where can I get some algorithms for DSP?
   2.2 What is 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?
   2.7 What is mu-law encoding?  Where can I get source for it?
   2.8 How can I do CD <-> DAT sample rate conversion?

3. Programmable DSP chips and their software.
   3.1 What are some current, popular programmable DSP chips?
   3.2 Software for Motorola DSPs.
    3.2.1 Where can I get a free assembler for the Motorola DSP56000?
    3.2.2 Where can I get a free C compiler for the Motorola DSP56000?
    3.2.3 Where can I get algorithms or libraries for Motorola DSPs?
          What is the number for the Motorola DSP BBS?
    3.2.4 Where can I get NeXT-compatible Motorola DSP56001 code?
   3.3 Software for Texas Instruments DSPs.
    3.3.1 Where can I get algorithms or libraries for TI DSPs?
                 What is the number for the TI DSP BBS?
    3.3.2 Where can I get a free C compiler for the TMS320C30?
    3.3.2 Where can I get a free assembler for the TMS320C30?
   3.4 Software for Analog Devices DSPs.
        3.4.1 Where can I get algorithms or libraries for AD DSPs?
          What is the number for the Analog Devices DSP BBS?

4. Hardware.
    4.1 DSP development boards.
    4.1.1 IBM PC.
    4.1.2 Mac Nubus.
    4.1.3 SBus.
    4.1.4 VMEbus.
    4.1.5 Next bus.
    4.1.6 Amiga bus.
    4.1.7 SCSI bus.
    4.1.8 TIM-40.
    4.1.9 Standalone.
    4.2 Who makes AES/EBU chips?

5. Operating Systems.

6. List of manufacturers, addresses, and telephone numbers.

7. Summary of FTP sites.

To search for a particular question within a text editor, look for the
pattern "Q#.##:", e.g., "Q1.2:".

=============================================================================

Q0: What is comp.dsp?

Comp.dsp is a worldwide UseNet news group that is used to discuss
various aspects of digital signal processing.  It is unmoderated,
though we try to keep the signal to noise ratio up :-).  If you need to
ask a question that isn't in the FAQ, and can't figure out how to post,
consult news.newusers.questions.  Other relevant news groups are
comp.compression, comp.speech, and sci.image.processing.

=============================================================================

1. General DSP.

This section deals with general DSP, that is, DSP books, algorithms,
and packages that are not related to a particular manufacturer's DSP chip.

=========================================================================

Q1.1: Summary of DSP books and significant research articles.

Q1.1.1: Bibles of DSP.

A.V. Oppenheim and R.W. Schafer, "Digital Signal Processing",
Prentice-Hall, Inc., Englewood Cliffs, N.J., 1975.

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.

L.R. Rabiner & R.W. Schafer, "Digital Processing of Speech Signals",
Prentice Hall, 1978, ISBN 0-13-213603-1.

R. E. Crochiere & L. R. Rabiner, "Multirate Digital Signal Processing",
Prentice-Hall, 1983, ISBN 0136051626.
This book is the only real reference for filter banks and multirate
systems, as opposed to being a tutorial.

P. P. Vaidyanathan, "Multirate Systems and Filter Banks",
Prentice-Hall.  911 pp.

Thomas Parsons, "Voice and Speech Processing", McGraw-Hill, 1987,
ISBN 0-07-048541-0.
Addresses the cocktail party effect, as well as other material.
[Maurice Givens, [e-mail]]

--------------------------------------------------

Q1.1.2: Adaptive signal processing.

S. Haykin, "Adaptive Filter Theory", 2nd Ed., Prentice Hall,
Englewood Cliffs, NJ, 1991.

B. Widrow and S.D. Stearns, "Adaptive Signal Processing",
Prentice-Hall, Inc., Englewood Cliffs, N.J., 1985.

--------------------------------------------------

Q1.1.3: Array signal processing.

J.E. Hudson, "Adaptive Array Principles", IEE London and New York,
Peter Peregrinus Ltd. Stevenage, U.K., and New York, 1981.

R.A. Monzingo and T.W. Miller, "Introduction to Adaptive Arrays"
John Wiley and Sons, New York, 1980.

S. Haykin, J.H. Justice, N.L. Owsley, J.L. Yen, and A.C. Kak
"Array Signal Processing", Prentice-Hall, Inc., Englewood Cliffs, N.J., 1985.

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 DFT",
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, [e-mail]]

Nezih C. Geckinli & 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.
[Bob Beauchaine, [e-mail]]

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, [e-mail]]

--------------------------------------------------

Q1.1.5: Digital Audio Effects Processing.

Books (in no particular order, sorry):

Hal Chamberlin, Musical Applications of Microprocessors, 2nd Ed.,
Hayden Book Company, 1985.

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.

Digital Signal Processing Committee of IEEE Acoustics, Speech, and Signal
Processing Society, ed. Programs for Digital Signal Processing. New York:
IEEE Press, 1979.

John Strawn, ed., "Digital Audio Signal Processing: An Anthology.", Los Altos,
CA: W. Kaufmann, 1985.   [Contains Moorer J.A. "About This Reverb..."
and contains an article which gives a code for Phase Vocoder -- great
tool for EQ, for Pitchshifter and more --Juhana Kouhia]

Charles Dodge and Thomas A. Jerse. Computer Music: Synthesis, Composition,
and Performance. New York: Schirmer Books, 1985.

F. Richard Moore, "Elements of Computer Music", Englewood Cliffs, NJ:
Prentice-Hall, 1990.  ISBN: 0-13252-552-6 [Recommended.  --Juhana Kouhia]

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]

John Strawn, ed., "Digital Audio Signal Processing", 283 pages,
$34.95, ISBN 0-86576-082-9, pub: A-R Editions.  Contents:

    1. Introduction to the mathematics of DSP (F. Richard Moore)
       [Not a bad little text]

    2. Introduction to digital filter theory (Julius O. Smith)
       [Not a bad little text, either]

    3. Spiral Synthesis (Tracy Lind Petersen)
       [first published account of a new synthesis technique]

    4. Signal processing aspects of computer music (J. A. Moorer)
       [James Moorer's classic article--discusses many synthesis
       techniques.  Reverb algorithms.  More than 6 pages of refs]

    5. An introduction to the phase vocoder (J. W. Gordon, J. Strawn)
       [Includes source code for a phase vocoder--a powerful method
       for synthesis, pitch shifting, time scale modification, etc.]
    [Comments by Quinn Jensen].

Curtis Road, ed., "Composers and the Computer", 201 pages,
$27.95, ISBN 0-86576-085-3, pub: A-R Editions.

John Strawn, ed., "Digital Audio Engineering", 144 pages,
$29.95, ISBN 0-86576-087-X pub: A-R Editions.

Deta S. Davis, "Computer Applications in Music: A Bibliography", 537 pages,
$49.95, ISBN 0-89579-225-7, pub: A-R Editions.

Ken C. Pohlmann, "The Compact Disc:  A Handbook of Theory and Use",
288 pages, $45.95 (cloth) ISBN 0-89579-234-6, $29.95 (paper)
ISBN 0-89579-228-1, pub: A-R Editions.

Forthcoming books:

Curtis Roads, "A Computer Music History:  Musical Automation
from Antiquity to the Computer Age"

Joseph Rothstein, "MIDI:  A Comprehensive Introduction"

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]

Check more articles from Journal of the Audio Engineering Society
(JAES), for example more articles by Strawn.

Note: books published by A-R editions can be ordered from:

    A-R Editions
    801 Deming Way
    Madison, Wisconsin 53717
    608-836-9000 (They accept VISA orders)

[The above is largely from Quinn Jensen, [e-mail];
Juhana Kouhia, [e-mail]; William Alves, [e-mail];
and Paul A Simoneau, [e-mail]]

=========================================================================

Q1.2: Where can I get free software for general DSP?

The packages listed below are mostly for general purpose DSP, that is,
DSP that is not specific to a particular programmable DSP chip.  See
the later sections in the FAQ for software relevant to a particular
programmable DSP chip.

According to Brian Evans: "There was an entire session on this subject
[free DSP software] at ICASSP '92, chaired by Dr. Sally Wood and
Dr. James McClellan.  It appears in Volume 4 of the Proceedings, pages
73-112.  There will be another such session at ICASSP '93." [Brian Evans,
[e-mail]]  Much of the information below is from Brian's mail.

--------------------------------------------------

Q1.2.1: What is Gabriel?  Where can I get it?

Package: Gabriel

Description:

Gabriel was a hierarchical block diagram environment for prototyping
signal processing systems on single or multiple processors.  It has
been superceded by Ptolemy (see below), and is no longer available.

--------------------------------------------------

Q1.2.2: What is Ptolemy?  Where can I get it?

Package: Ptolemy

Description:
Ptolemy provides a highly flexible foundation for the specification,
simulation, and rapid prototyping of systems.  It is an object
oriented framework within which diverse models of computation can
co-exist and interact.  For example, using Ptolemy a data-flow
system can be easily connected to a hardware simulator which in turn
may be connected to a discrete-event system, etc.  Because of this,
Ptolemy can be used to model entire systems.

In addition, Ptolemy now has code generation capabilities.  From a
flow graph description, Ptolemy can generate both C code and DSP
assembly code for rapid prototyping.  Note that code generation is
not yet complete, and is included in the current release for
demonstration purposes only.

Ptolemy has been used for a broad range of applications including
signal processing, telecomunications, parallel processing, wireless
communications, network design, radio astronomy, real-time systems,
and hardware/software co-design.  Ptolemy has also been used as a lab
for signal processing and communications courses.  Currently Ptolemy
has hundreds of users in over 100 sites, both in industry and academia.

Ptolemy is available for the Sun 4 (sparc), DecStation (MIPS), and HP
(HP-PA) architectures. Installing the system requires 49 Mbytes for
Ptolemy (64 Mbytes after you optionally rebuild) and 16 Mbytes for the
Gnu tools subset.  At least 8 Mbytes of physical memory are required.

Ptolemy has been developed at UC Berkeley over the past 3 years.
Further information, including papers and the complete release
notes, is available from the FTP site.

A license is no longer required to receive Ptolemy.  The source code,
binaries, and documentation are available by anonymous ftp from

    ptolemy.berkeley.edu

Ftp to "ptolemy.bekeley.edu", login as "anonymous", cd "pub", get README

Contact: Alan Kamas, [e-mail].
--------------------------------------------------

Q1.2.3: What is Khoros?  Where can I get it?

Package: Khoros

Description:
Block diagram simulator for image and video processing.  1-D signal processing
is also supported.  See the UseNet group comp.soft-sys.khoros.

Platforms: sun 3, sun 4, others?  X windows.  Written in C.

To obtain:
Anonymous ftp to pprg.eece.unm.edu, cd /pub/khoros/release, get install.ftp.
--------------------------------------------------

Q1.2.4: What are DSP Tutorials?  Where can I get them?

Package: DSP Tutorials

Description: Computer aided instruction.

Platforms: suns under SunView.

Contact: Dr. Sally Wood, Electrical Engineering Department, Santa Clara
University, Santa Clara, CA 95053.
--------------------------------------------------

Q1.2.5: What are some DSP extensions to MATLAB?  Where can I get them?

Package: 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  make
    submissions  to the digest, please send  to [e-mail] with a
    subject: "DIG" and description.

    For  the  Pacific,   try  [e-mail]  located   at  the
    University of Wollongong, NSW, Australia.

    A plethora of toolboxes are available at FTP site: research.att.com
    (use netlib for the username)

    General index for the MATLAB User Group software library
    ----------------------------------------------------------
    Currently there are the following subdirectories:
    approximation      approximation theory
    archive            old MATLAB user group digests
    control            control theory
    dataanalysis       data analysis and statistics
    graphics           graphics programs
    integration        numerical integration
    linearalgebra      linear algebra utilities
    misc               miscellaneous
    ode                ordinary differential equations
    optimization       as the name says
    pde                partial differential equations
    rootfinding        zero-finding routines
    specialfunctions   special functions
    teaching           for classroom use
    tools              miscellaneous tools

    In order to get an index for a subdirectory (tools, say)
    send the message
        send index from MATLAB/tools
    to [e-mail].

    In order to get some code, (unbundle in the `tools'
    directory, say), send the message
        send unbundle from MATLAB/tools
    to [e-mail].

    FOR  STUDENTS: Prentice Hall has published  a student edition of matlab
    which contains a book and  set of disks for PCs and Macs.  The software
    is limited only in matrix size (32 x 32 matrix;  1024 elements) and  in
    its ability  to import or call  C or Fortran  subroutines.  On the plus
    side, it is able to run without a coprocessor (it will use one if it is
    present) and it includes a subset of the Signal Processing and Controls
    Toolboxes, The Signals and  Systems Toolbox,  which  provides for added
    functionality. Book only (about US$30): ISBN = 0138560064; Book + disk:
    (about US$50 ISBN=0-13-855974-0 for 3.5" or ISBN=0-13-855982-1 for 5.25
    Macintosh version:  ISBN=0-13-855990-2. There will be related books out
    by mid to late 1993 :  Computer Aided Signal Processing with MATLAB, by
    Burrus, Oppenheim, McClellan, Parks, Schafer, and Schussler; and Signal
    Processing : A Computer  Approach, by Etter. More books in this  MATLAB
    Curriculum Series are planned. For general info: [e-mail]

    [From the Matlab Users Group (Editor, [e-mail])]
--------------------------------------------------

Q1.2.6: What are the Signal Processing Packages for Mathematica?
        Where can I get them?

Package: Signal Processing Pacakages (SPP) and Notebooks.

Description:

         Public domain  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, and number theoretic operations  (Bezout
    numbers, Smith Form decompositions, and matrix factors).   Accompanying
    SPP  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.  A Notebook reader is available in the public domain
    for Macintosh computers (a BinHexed version is on the ftp site).

    FOR STUDENTS:   A student version of Mathematica is available for $175.
    The price includes a copy of the reference manual.  The only drawbacks
    to the student version are that the floating point coprocessor is dis-
    abled and that upgrades cannot be ordered.

Contact: Brian Evans, [e-mail].

To obtain: anonymous ftp to gauss.eedsp.gatech.edu (130.207.226.24).
(available in UNIX, Mac, and MS-DOS archive formats).

--------------------------------------------------

Q1.2.7: What is the Control Systems Analysis Packages for Mathematica?
        Where can I get them?

Package: 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).

Contact: Dr. Sreenath, [e-mail].

To obtain: anonymous ftp veda.esys.cwru.edu (129.22.40.9).
--------------------------------------------------

Q1.2.8: What are some other Mathematica DSP Notebooks?

The following Mathematica notebooks can be ftped from ccrma-ftp.stanford.edu:

pub/DSP/GenHamming.ma.Z         Generalized Hamming windows
pub/DSP/Kaiser.ma.Z             The Kaiser window
pub/DSP/WinFlt.ma.Z             Digital filter design by the "window method"

(There are other DSP related items in pub/DSP on ccrma-ftp; see other
sections of this FAQ for details).
--------------------------------------------------

Q1.2.9: What is the Linear Systems Toolbox for Maple?
        Where can I get it?

Package: Linear systems toolbox for Maple.

Description: Public domain extension to Maple.

Contact: Tony Richardson, [e-mail].

To obtain: anonymous ftp to ftp.egr.duke.edu
file pub/maple/linsys1.2.tar.Z.
--------------------------------------------------

Q1.2.10: Where can I get text to speech conversion software?

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, [e-mail]]
--------------------------------------------------

Q1.2.11: Where can I get filter design software?

There are filter design programs available via anonymous FTP.  The
following are summarized here and discussed in greater detail below:

1. August 1992 IEEE Trans. on Signal Processing: METEOR FIR filter design
   program.

2. DFIR FIR filter design program.

3. Netlib IIR filter design.

4. IEEE Press "Programs for Digital Signal Processing".

-----

The August 92 issue of IEEE Transactions on Signal Processing there is
a paper entitled "METEOR: A Constraint-Based FIR Filter Design Program"
by Kenneth Steiglitz, Thomas W. Parks and James F. Kaiser.  They
describe an FIR design program which allows specification of the target
frequency response characteristics in a fairly generalised and flexible
way.  As well as designing filters, the program can optimise filter
lengths and push band limits.

The paper contains a footnote which says "Pascal and C versions of
source code are available to  anonymous ftp at princeton.edu in the
directory /pub as meteor.p, form.p, meteor.c and form.c".

True, they are.  They appear to work.  The Pascal versions have been
put through p2c to get the C versions; all the needed Pascal library
stuff is included in the C versions and they built error-free out of
the box for me on an SGI machine.

One catch is, there is no manual - you need the paper to know how to drive
the programs.

[Steve Clift, [e-mail]]

-----

Another public domain filter design package is DFIR, for FIR filter designs.
It includes design capabilities for: equiripple linear phase multiband filters,
linear phase differentiators, linear phase Hilbert transform filters, MMSE
interpolating filters and equiripple Nyquist filters.  It is written in
Fortran 77 and has been tested on DECStations and Suns.

It is available via anonymous ftp from aldebaran.ee.mcgill.ca in "pub/dfir".
Additionally, a package to plot filter responses is available in
"pltfilter-V2R0.tar.Z".

[Peter Kabal, via Witold Waldman]

-----

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.
research.att.com) as the file netlib/cephes/ellf.shar.Z.  By email to
[e-mail] the request message text is `send ellf from cephes'

[Stephen Moshier, [e-mail]]

-----

The Fortran source code from the IEEE Press book "Programs For
Digital Signal Processing" is available for anonymous ftp from
nimios.eng.mcmaster.ca in directory pub/IEEE/software as dsp.zip.
It includes FIR and IIR filter design software, as well as other
general purpose DSP subroutines.

There is also a C/C++ version of the Parks-McLellan FIR filter
design program available for anonymous ftp from ftp.uu.net in
directory /usenet/comp.sources.misc/volume22/fir as file part01.Z.
This program was created and tested using Borland C++ 2.0.

[Witold Waldman, [e-mail], from
Charles Owen at [e-mail]]

{ There are other free filter design programs floating around out there,
  such as optfir/wfir.  Does anyone know of ftp sites? }

--------------------------------------------------------------------------
Q1.2.12: What is SigLib?  Where can I get it?

SigLib is an ANSI C Source Library of DSP functions that has been used
on a variety of platforms containing both General purpose micros and
DSPs.  Siglib contains graphical front ends for DOS and Windows, the
full C Source, example programs and registered users get one years free
upgrade, maintenance and support.  SigLib is distributed as shareware and
as a commercial product.  The shareware version is available in object code
only, however.

The following is a brief overview of some of the functionality of
SigLib: spectrum analysis (including frequency zoom capabilities),
windows, fixed, multirate (polyphase) and adaptive filters, filter
design methods, convolution (1 and 2 dimensions), correlation, signal
generation, modulation and a whole suite of DSP utility functions and
graphic routines.

Contact: John Edwards, [e-mail].

To obtain: Compuserve, or 157 Sileby Road, Barrow-on-Soar, Leics, LE12 8LW, UK.

{If someone downloads this, mail a note to Brad Hards, [e-mail],
and he will put it up for anonymous FTP.}

--------------------------------------------------------------------------
Q1.2.13: What is the AudioFile System?  Where can I get it?

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.

AudioFile is distributed in source form, with a copyright allowing
unrestricted use for any purpose except sale (see the Copyright notice).
[e-mail] is a mailing list for discussions of AudioFile. Send
mail to [e-mail] to be added to this list.

The kit is located at ftp site crl.dec.com (Internet 192.58.206.2) in
/pub/DEC/AF.  The kit is contained in a compressed tar file named
AF2R2.tar.Z.  Use anonymous ftp to retrieve the file.
AF2R2-other.tar.Z is a sample kit of Hi-Fi sound bites, if you
have suitable hardware such as DECaudio.

[Larry Stewart, [e-mail]]

=============================================================================

2. Algorithms and standards.

This section deals with DSP algorithms and related standards.

=======================================================================
Q2.1: Where can I get some algorithms for general DSP?

The following archives contain things such as matrix operations, FFT's and
generally useful things like that, as opposed to complete applications:

(1) host ux1.cso.uiuc.edu - log in as anonymous, get the file math/README.
    It summarises the contents of the archive.

(2) Netlib, which serves some of this software via email. Try mail to
    [e-mail] with "send help" in the subject field.
    For Europe:
        Internet:       [e-mail]
        EARN/BITNET:    [e-mail]
        X.400:          s=netlib; o=nac; c=no;
        EUNET/uucp:     nac!netlib
    For the Pacific, try
        [e-mail]
     For background about netlib, 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
           [e-mail].

     The symbolic algebra system REDUCE is supported by
          [e-mail].

(3) The Naval Surface Warfare Center has a library of mathematical
    Fortran subroutines that may be of use.  From the report itself:

    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.

    Distribution: Approved for public release; distribution unlimited.

    Abstract:

    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

    [Witold Waldman, [e-mail]]

    Apparently euler.math.usma.edu has these routines in the file
    "pub/misc/nswc.tar.Z".  This is a 3.2 Mbyte file with 800+ Fortran
    routines mentioned above.  Apparently euler.math.usma.edu has been
    down a lot lately, however, and the network connection to it is
    quite slow.  If somebody would like to make this available from
    a more reliable site, please let us know at [e-mail].

(4) The Fortran source code from the IEEE Press book "Programs For
    Digital Signal Processing" is available for anonymous ftp from
    nimios.eng.mcmaster.ca in directory pub/IEEE/software as dsp.zip.
    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.

    [Witold Waldman, [e-mail], from
    Charles Owen, [e-mail]]

Also, see the summary of DSP-related FTP sites, at the end of this FAQ.

If you don't know where to find what you're after, try archie.

=======================================================================

Q2.2: What are CELP and LPC?  Where can I get the source for CELP and LPC?

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 (FS 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
at no charge (on DOS diskettes, but configured to compile on Sun SPARC
stations) from:

    Bob Fenichel
    National Communications System
    Washington, D.C.  20305
    1-703-692-2124
    1-703-746-4960 (fax)

Example input and processed speech files, a technical information bulletin,
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 included at no charge.

Unfortunately, a document that is a vital part of the CELP release package
is not available in electronic form.  Anyone serious interested in CELP
should obtain the document:

    Details to Assist in Implementation of Federal Standard 1016 CELP.
    National Communications System, Office of Technology & Standards, 1992.
    Technical Information Bulletin 92-1.

It is available from the above address.

The following article describes the FS 1016 4.8-kbps CELP coder:

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.

The U.S. DoD's Federal-Standard-1015/NATO-STANAG-4198 based 2400 bps
linear prediction coder version 53 (LPC-10e v53) Fortran or C simulation
source codes are available on a limited basis upon written request to:

    Tom Tremain
    Department of Defense
    Ft. Meade, MD  20755-6000
    USA

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, p. 40-49.

Copies of the official standards FS 1015 and FS 1016 are available for
US $2.50 each from:

    GSA Rm 6654
    7th & D St SW
    Washington, D.C.  20407
    1 [phone]


Realtime DSP code for FS-1015 and FS-1016 is sold by several vendors,
including DSP Software Engineering and Analogical Systems (see the
vendor address list in section 5 for contact info).  DSP Software
Engineering's FS-1016 code can run on a DSP Research's Tiger 30 or on
Intellibit's AE2000 TMS320C31 based 3" by 2.5" card.  See section 4.1
for more on these cards.  Analogical's product runs on a 27 MHz
DSP56001 chip.

[Most of the above from Joe Campbell, [e-mail], with
additions from Dan Frankowski, [e-mail], and Ed Hall,
[e-mail]]

{The CELP code is no longer available for FTP.  If anyone knows of a
new ftp site, please send mail to [e-mail].}

=======================================================================
Q2.3: What is ADPCM?  Where can I get source for it?

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 samples and converts
them to 4-bit samples, yeilding a compression rate of 4:1.

There is public domain C code available via anonymous ftp at
ftp.cwi.nl in /pub/adpcm.shar written by Jack Jansen (email
[e-mail]).  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);

The routines have been tested on an SGI Indigo running Irix 4.0.2 and
on a Sparcstation 1+ running SunOS 4.1.1.  On a Sun, the code will
compress at 250Ksample/sec and decompress at 300Ksample/sec.  On an
SGI, the compressor runs at 350Ksample/sec and the decompressor at
700Ksample/sec.

Note that this is NOT a CCITT G722 coder.   The CCITT ADPCM standard is
much more complicated, probably resulting in better quality sound but
also in much more computational overhead.

[From Dan Frankowski, [e-mail]; Jack Jansen, [e-mail]]

=======================================================================
Q2.4: What is GSM?  Where can I get source for it?

The README file for GSM says:

GSM 06.10 13 kbit/s RPE/LTP speech compression available
--------------------------------------------------------

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 we 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 modelled after compress(1), and
a library API.  Compression and decompression run faster than realtime
on most SPARCstations.  The implementation has been verified against the
ETSI standard test patterns.

Jutta Degener ([e-mail])
Carsten Bormann ([e-mail])

Communications and Operating Systems Research Group, TU Berlin
Fax: [phone], Phone: [phone]

An implementation can be FTPed from
    tub.cs.tu-berlin.de:/pub/tubmik/gsm-1.0.tar.Z
                           +/pub/tubmik/gsm-1.0-patch1
                           +/pub/tubmik/gsm-1.0-patch2

or as a faster but not always up-to-date alternative:
    liasun3.epfl.ch:/pub/audio/gsm-1.0pl1.tar.Z

[From Dan Frankowski, [e-mail]]
============================================================================
Q2.5: How does pitch perception work, and how do I implement it on my DSP chip?

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.

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, 1989.

This book is available in paperback and makes a good desk reference.

An algorithm implementation that matches a large body of psychoacoustical
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.

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, Apple Computer, and John Lazzaro,
U.C. Berkeley.]

=======================================================================

Q2.6: What standards are there for digital audio?  What is AES/EBU?
      What is S/P-DIF?

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 raq CD subcode; DAT is StartID and
SkipID.  In progfessional mode, there is an SDLC protocol or, if DAT,
it may be the same as consumer mode."

There 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" (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, [e-mail], and Tim Channon,
[e-mail]]

=============================================================================

Q2.7: What is mu-law encoding?  Where can I get source for it?

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.

Craig Reese posted the source of similar routines to comp.dsp in August '92.
These are archived on evans.ee.adfa.oz.au (131.236.30.24) in /pub/dsp/misc

References:

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",
pp. 169-198.

[From Joe Campbell; Craig Reese, [e-mail]; Sepehr Mehrabanzad,
[e-mail]]

=============================================================================

Q2.8: How can I do CD <-> DAT sample rate conversion?

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, [e-mail]:]

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.

In any case, your local DSP guru will probably be able to suggest a good
program for designing digital filters (this is a good general question
for the net, anyway).  This is one case where going beyond the simpler
windowed-sinc-based approaches might pay off (and I'm afraid I can't
recommend any readily available programs myself).

=============================================================================



3. Programmable DSP chips and software.

This section deals with programmable DSP chips and their software.

Q3.1: What are the available DSP chips and chip architectures?

{ This is based on a woefully inadequate databook collection.  Anyone want
  to add to this list?  Manufacturers want to submit anything? }

The "big four" programmable DSP chip manufacturers are Texas Instruments,
with the TMS320 series of chips; Motorola, with the DSP56000 and DSP96000
series; AT&T, with the DSP16 and DSP32 series; and Analog Devices, with
the ADSP2100 series.  A good overview of prorammable DSP chips is published
periodically in EDN magazine.  The most recent version is from Sep. 17, 1992,
pp. 90-141.

Here's a less ambitious chip breakdown by manufacturer:

-------------------------------------

Texas Instuments:

TMS320C1x: family of low cost fixed-point DSP's; 16 bit data, 32 bit
registers; Various RAM and ROM configurations; 16 bit I/O bus, serial ports.

TMS320C25: 50MHz fixed-point DSP; 16 bit data, 32 bit registers;
12.5 MIPS @ 50MHz.

TMS320C30: 27/33/40 MHz floating point DSP; 32 bit floating point,
24 bit fixed point data, 40 bit  registers; DMA  controller; dual serial
ports; some support for multi-processor arrays.

TMS320C31: version of C30 minus peripheral bus, one serial port, and the 4Kx32
internal ROM.  ~$20, 132 pin PQFP.

TMS320C40: 40/50 MHz floating point DSP; extensive parallel processing
support through 6 buffered byte-wide 20 Mb/s links and 6 channel DMA; cache.

TMS320C50: enhanced TMS320C25 (double throughput); low overhead looping;
10 Kwords SRAM on chip.

-------------------------------------

Motorola:

DSP56001: 20.5, 27, or 33 MHz 24-bit fixed point DSP.  24 bit data bus, 16 bit
address bus, 56 bit accumulators (2), host interface port, serial ports (2),
general purpose I/O pins, timer.  Harvard architecture.  512 words program
RAM, 32 words bootstrap ROM, 512 words data RAM, 512 words data ROM on chip.
Available in PGA, CQFP or PQFP packaging.

DSP56000: Mask-programmed version of DSP56001, same peripherals and data
memories, 3.75k words program ROM on chip.

DSP56002: modular DSP based on new 24-bit DSP56k core, a superset of
the DSP56001 architecture with On-Chip Emulation (OnCE) debug port,
clock PLL and improved bus arbitration. Has four cycle double precision
multiply and support for block floating point. Same memory as in
DSP56001, except for 64 words bootstrap ROM.  Host interface port,
serial ports (2), general purpose I/O pins, programmable 24-bit timer,
non-maskable interrupt.  Low power fully static design, no minimum
clock frequency requirement.  Available at 40 MHz (5V supply) in PGA
and CQFP packaging.

DSP56004: modular DSP, same 24-bit DSP56k core as in DSP56002. Targeted
to consumer digital audio applications.  Has On-Chip Emulation (OnCE)
debug port, clock PLL, serial host interface (I2C and SPI), four
general purpose I/O pins, two stereo serial audio receivers (I2S/Sony),
three stereo serial audio transmitters (I2S/Sony), external SRAM/DRAM
memory interface with 8-bit data bus.  Low power fully static design,
no minimum clock frequency requirement.  Available at 40 MHz (5V
supply) in 80-pin QFP package.

DSP56156: 40, 50, or 60 MHz fixed point DSP; 16 bit data bus, 40 bit
accumulators (2), host interface port, serial ports (2), timer, OnCE
debug port, clock PLL, 14 bit sigma-delta voice band CODEC, 2K words
program RAM, 2K words data RAM on chip.

DSP96002: IEEE format floating point DSP; two complete 32 bit data and
address buses; Harvard architecture. 1k words program RAM, 64 words
bootstrap ROM, 1k words data RAM, 1k words data ROM, host interface
ports (2).  Available in 33 MHz or 40 MHz in 223-pin PGA packaging.
[The above from   Sergio Liberman, [e-mail]    ]

-------------------------------------

AT&T:

DSP 16 FAMILY: DSP16A, $22.60; DSP1610, $91; DSP1616, $35.70
16 bit fixed point DSPs.  The DSP16A has a 25ns cycle time while the
16C has an ADC and DAC on chip. The C-version also has a 4-pin JTAG
interface.  The 1610/1616 are enhanced versions intended for cellular
phone use.  The chips use separate on-chip 16-bit program- and data
buses.  The A and C versions have 12k x 16-bit program ROM and a 2k x
16-bit data-RAM while featuring parallell and serial I/O.

DSP 32C/3210: DSP32C, $70(1000); DSP3210, $50(100k)
32-bit floating point with 40-bits accumulator and 16/24-bit fixed
point. These DSP's uses a single 4M-word (3210: 4G-word) linear memory
space instead of the separate program and data memory found on most
DSP:s. The bus can be accessed four times and each internal memory two
times per cycle. The 3210, along with the VCOS operating system is
intended for use on the mother board of personal computers and
workstations where it shares memory with the host.  The 32C has three
512 x 32-bit RAM:s while the 3210 has two 1k x 32-bit RAMs and a 256 x
32-bit boot ROM. 32C: Serial and parallell I/O, 3210: Serial I/O,
timer, DMA-controller.  3210 available at 50 and 66MHz.

-------------------------------------

Analog Devices:

ADSP-2100:  40 and 50Mhz fixed point DSP (10 MIPS, 12.5MIPS).  16 bit
operations with 40-bit multiply-accumulate.  No on chip memory except for a 16
word instruction cache.  Off chip harvard architecture.  PGA and PQFP
packages.

ADSP-2101:  Derived from ADSP-2100; 16 bit operations with 40-bit MAC
register.  Extras include on chip memory of 2Kx24 program memory
instruction/data RAM and 1Kx16 data memory RAM, 16-bit timer, two serial
ports, low power state.  PGA, PLCC, PQFP packages.  Fastest speed grade in
production is 16.6 MHz (16.6MIPS).

ADSP-2102:  RAM/ROM version of 2101; user selects how much of the 2kx24
program memory is mask ROM.

ADSP-2103:  3.3V version of the 2101 running at 13MHz (13 MIPs).  PLCC, PQFP
packages.

ADSP-2105:  10Mhz (10MIPS) low cost fixed point DSP with 1 serial port, timer
and 1kx24 instruction/data RAM in program memory space, and 512 word data RAM
in data memory space.  Architecture and instruction set identical to
ADSP-2101.  Pin compatible with 2101.  PLCC package only (can use standard 68
pin plcc socket).  This processor sells for US $9.90 in any quantity.

ADSP-2115:  Architecture and pinout same as ADSP-2101, but 1K program memory
RAM, 1/2K data memory RAM with 2 serial ports, interval timer etc.  PLCC and
PQFP packages.  Available in 13.8MHz and 16MHz (13.8, 16 MIPs)

ADSP-2111:  Adds a 8/16bit host interface port to ADSP-2101 architecure
allowing interface to Intel or Motorola style microprocessors.  13 and 16 MIPs
speeds available.  PGA and PQFP packages.

ADSP-21msp50:  ADSP-2111 with an on chip a/d and d/a interface (65dB SNR)
Additional low power modes allow CMOS standby (<100uA).  100-lead PQFP
package.

ADSP-216x:  Four devices, two with 8Kx24 mask ROM and two with 2K*24 mask ROM,
each has 1/2K data memory RAM.  Pin-for-pin compatible with ADSP-2101.  PLCC,
PQFP packages.

ADSP-21020:  20/25/33 MHz floating-point DSP; Supports 32-bit fixed point,
IEEE format 32-bit floating point, and 40-bit floating point; 40-bit registers
plus two 80-bit fixed-point multiply-accumulators; Harvard arch.  with 32 word
instruction cache allows two data accesses in a single cycle; IEEE 1149.1 JTAG
boundry scan; 33.3 MIPS @ 33.3 MHz.

ADSP21010:  Slower inexpensive version of '020 (16 MHz).  Limited to 32-bit
fixed and floating point.

All of the processors (except the 2100) use a 1X instruction clock and use an
on chip PLL to generate an internal 4X clock.  All processors have an extended
Harvard architecture which allows two data fetches and an instruction fetch
every cycle in parallel with an alu or mac operation.

Processors with internal memory support booting from inexpensive 8 bit EPROMS.
All processors include hardware support for zero overhead looping, modulo
addressing, single cycle context switch, and bit reversal addressing.  All
instructions, even those which access external memory can complete in 1 cycle.

[Analog Devices DSP applications, [e-mail]]

=======================================================================

Q3.2: Software for Motorola DSPs.

=======================================================================

Q3.2.1: Where can I get a free assembler for the Motorola DSP56000?

A free assembler for the Motorola DSP56000 exists, thanks to Quinn
Jensen, [e-mail].  The current version is 1.1, and
it is posted to alt.sources, so look for it on mirrors of that
newsgroup (like wuarchive.wustl.edu).

=======================================================================

Q3.2.2: 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 ([e-mail]) 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 Q3.2.1 above).

The Motorola gcc source is available for FTP from:

    nic.funet.fi        ~pub/ham/dsp/dsp56k-tools/dsp56k-gcc.tar.Z
    evans.ee.adfa.oz.au    pub/micros/56k/g56k.tar.Z

From Andrew Sterian, [e-mail]:

My DSP56156 port is still the only DSP56156 compiler around and I have
just released an updated version of it.  Both this compiler and the
previous incarnation are archived on wuarchive.wustl.edu (in the
usenet/alt.sources directory) amongst other places.

=======================================================================

Q3.2.3: Where can I get algorithms and libraries for Motorola DSPs?
    What is the number for the Motorola DSP BBS?

Motorola runs "Dr. Bub", a bulletin board for DSPs containing source code
for various libraries and algorithms.  You can call it at [phone]
(9600, 4800, 2400, 1200 bps) or [phone] (2400/1200/300 bps).
Format is 8 data bits, no parity, 1 stop bit).  Log in as "guest" to
browse the system, or you can open an account by entering "new" at the
account name prompt.  [John Fisher, [e-mail]]

Alternatively, Dr. BuB is mirrored on the following sites:

calvin.stanford.edu   (36.14.0.43)     in   /motorola
bode.ee.ualberta.ca   (129.128.16.96)  in   /pub/dos/motorola
nic.funet.fi          (128.214.6.100)  in   /pub/misc/motorola
doc.ic.ac.uk          (146.169.3.7)    in
/computing/systems/motorola/digital-signal-processing/dr.bub.sources

Also try nic.funet.fi in /pub/ham/dsp for a lot of good stuff on
communications uses, including some hardware.

ccrma-ftp.stanford.edu also has a variety of DSP code (much of it NeXT
specific, see below), including the following for the DSP56000:

pub/clm.tar.Z        "CLM", a package aimed mainly at composers doing
            computer music in Common Lisp, but includes a Lisp
            56000 assembler, debugger, loader, large libraries of
            DSP56000 routines useful in computer music, and
            a compiler from a subset of Common Lisp to
            DSP56000 code.  [[e-mail]]

=======================================================================

Q3.2.4: Where can I get NeXT-compatible Motorola DSP56001 code?

Try the following from ccrma-ftp.Stanford.EDU:

DSP programs for the NeXT platform:

pub/DSP/resample.tar.Z          Audio sampling-rate conversion and FIR
                filter design.
pub/DSP/ResoLab2.1.tar.Z        Interactive filter instrument; sources now
                included, online help.
pub/DSP/Spectro.Z               Spectrum analysis tool, with source code.
pub/DSP/WaveFormEditor.tar.Z    Jean Laroche's real-time waveform editor,
                with DAJ's additions.

DSP programming examples for the NeXT platform:

pub/DSP/dsp_dma_stream.tar.Z    Fast DSP DMA programming example (two-way DMA).
pub/DSP/JeanLaroche.tar.Z       Low-level sound and DSP programming examples
                and docs.

[[e-mail]]

=======================================================================

Q3.3: Software for Texas Instruments DSPs.

=======================================================================

Q3.3.1: Where can I get free algorithms or libraries for TI DSPs?
        What is the number for the TI DSP BBS?

nic.funet.fi has some old, apparently public domain, assembler and
related tools from TI for the TMS320 family.  [Antti-Pekka Virtanen,
[e-mail]]

The TI DSP bulletin board is at [phone] (300, 1200, 2400, or 9600 bps;
8 data, 1 stop, no parity).

evans.ee.adfa.oz.au has a mirror of the TI DSP bulletin board in
"mirrors/tibbs".  See the "00README" file at the top of the
directory tree for info.  Please restrict FTP session to outside of
8 am to 6 pm local time (10 pm to 8 am GMT).
This is mirrored from ti.com, in /tms320bbs. The TI site is
the official one, but has no user contributed software.
[Brad Hards, [e-mail]]

{ If anyone knows of any other sources for TI DSP software, please let
us know at [e-mail].  Thanks! }

=======================================================================

Q3.3.2: Where can I get a free C compiler for the TI TMS320C30?

Sonitech (see vendors list) has a gcc based TMS320C30 C compiler
that was originally done Computer Motion.  Sonitech sells it
for $995, but under the terms of the Gnu Public License, other
people can then give it away.  While we haven't heard of any
ftp sites yet, there are bound to be some soon.

=======================================================================

Q3.3.3: Where can I get a free assembler for the TI TMS320C30?

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)."

It is available for anonymous ftp from schutz.ee.uts.edu in
~/pub/DSP/c30/as30.tar.Z".

[Ted Rossin, [e-mail]]

=======================================================================

Q3.4: Software for Analog Devices DSPs.

=======================================================================

Q3.4.1: Where can I get algorithms or libraries for Analog Devices DSPs?
        What is the number for the Analog Devices DSP BBS?

The number for the Analog Devices DSP BBS is [phone] (300, 1200,
2400, 9600, 14400 bps), 8N1.

[Analog Devices DSP Applications, [e-mail]]

{ If anyone knows of other sources for Analog Devices DSP software,
please let us know at [e-mail].  Thanks! }

=======================================================================

Questions, comments, or submissions for this FAQ should be mailed to
[e-mail], ...!ucbvax!ohm!comp-dsp-faq.  Thanks!