HDTenEightyP/NetNews huggingface dataset:
From: rl5499@ehsn21.cen.uiuc.edu (Robert M Laskey III)
Subject: 56001 C compiler
Summary: Looking for Moto56001 C compiler and assembler.
Organization: University of Illinois at Urbana
Date: Tue, 6 Oct 1992 20:48:47 GMT
Keywords: Motorola DSP 56000 C compiler

I need a quality C compiler for Motorola's 56K line of DSP's. The
development SW I have includes a monitor, which is adequate for now but
its usefulness will soon be gone. Any information on C comilers and/or
assemblers for this processor would be greatly appreciated.

Please respond by mail and I'll post a summary if anyone else is interested.

thanks.

| Bob Laskey
| University of Illinois, Urbana-Champaign   rml31294@uxa.cso.uiuc.edu
| Electrical and Computer Engineering
|       		*VOTE BUSH/QUAYLE in 1992*
From: chuckc@hpnmdla.sr.hp.com (Chuck Corley)
Date: Mon, 5 Oct 1992 16:55:21 GMT
Subject: Re: Looking for DSP op system feedback
Organization: Hewlett Packard, Santa Rosa, CA
Newsgroups: comp.dsp


        I've been asked to post the phone numbers and addresses for
companies that make operating systems that run on DSP processors.  Here
are the ones I know of:


'Nucleus RTX' is available from:

  Accelerated Technology
  P.O. Box 850245
  Mobile, Alabama 36685, USA

  Phone:     (205) 661-5770
  Toll free: (800) 468-NUKE
  Fax:       (205) 661-5788


'SPOX' is available from:

  Spectron Microsystems
  5266 Hollister Ave
  Santa Barbara, CS 93111, USA

  Phone:     (805) 967-0503
  Fax:       (805) 683-4995


        If you call these people and you don't mind doing this, please
mention that you heard about them "from a posting by Chuck Corley in the
note string 'comp.dsp'".  I think this will help raise the visibility of
'comp.dsp' as a DSP communications medium.  This will also help me while
I'm trying to get information from them in order to do my evaluation of
the two operating systems.

        Perhaps in the future these companies will monitor and participate
in 'comp.dsp' as a result.  (I've noticed that some of the leading DSP
suppliers have already done this).  Who knows, someday they may even
change a critical policy or add some needed feature as a result ...

	Hope that this helps those who requested the information.

    					Chuck Corley
    					chuckc@sr.hp.com

The standard disclaimer:
The opinions and comments expressed here are mine and not those of my employer.
Newsgroups: comp.dsp,sci.image.processing
From: dak@kaa.informatik.rwth-aachen.de (David Kastrup)
Subject: Re: Fourier Transform, Ringing
Organization: Rechnerbetrieb Informatik  /  RWTH Aachen
Date: 26 Sep 92 08:56:02 GMT

kr0u+@andrew.cmu.edu (Kevin William Ryan) writes:

>patel@sparky.drad.umn.edu
>>       I need some help in terms of some ideas or pointers
>>       to literature. The problem is as follows:
>>       I have two signals (complex) which are Fourier transform
>>       of an object of interest. Call them x1 and x2. x1 contains
>>       only low frequencies where as x2 has low+high frequencies.
>>       However, x2 is corrupted by additive noise (can be appro-
>>       ximated by white Gaussian)
>>
>>       This problem can be simulated as follows:
>>       Take a 64 point DFT of a time signal(could be
>>       complex or real). Then x1 corresponds to 16
>>       points of this DFT centered around the dc value
>>       and x2 corresponds to the full 64 point DFT
>>       with additive noise.
>>
>>       I would like to reconstruct my object object from
>>       this two sets of data. If I use x1 and zero pad it, I
>>       get ringing in my reconstructed signal near sharp
>>       peaks. To avoid this ringing, I want to somehow use the
>>       hig frequency data that I have (signal x2). I am trying
>>       out somethings like filtering the x2 signal and adding
>>       etc etc. I need some pointers to any work done in this
>>       directions or if anybody has a good solution to this
>>       even some ad hoc solution will do.

>    All right, here's an ad hoc solution for you...

>    If X1 is the true uncorrupted low frequencies of your signal and X2 is
>the Fourier of the entire signal plus noise, you can find out what your
>noise is (approximately) by:

>        (X2 * lowpass) - X1 = model of noise

>    If the noise (as you seem to be saying) is white noise you now have
>an average level of the noise, and can subtract that from X2 to give a
>better estimate of te signal (Xs). If the noise is Gaussian in frequency
>space you again have a model of the amplitude of the noise and can (with
>a little more difficulty, to be sure) estimate it's size elsewhere.

>    To do this (warning, this is off the top of my head) you should
>reduce X2 by the average magnitude of the noise model.

>    This is, however, dependent on the amplitude of X1 being identical
>with the amplitude of X2 - no scaling.

>    Given a good model of the noise magnitude, I would subtract it from
>X2, substitute X1 for the low frequency components of X2, and use that as
>the model for your signal Xs.
Uh oh. You forgot that although gaussian noise has pretty constant
amplitude, its phase is COMPLETELY random and might in ANY way
disturb the signal.

Thje best approach, if you really take the lower data to approximate
the error, is to use x1 where it is nonzero (NO error at all),
and to use S/(S+N) as a factor elsewhere, where N is the estimated
noise ENERGY in the spectre, and S is the expected signal energy.

This will cause the lowest noise energy (Wiener filter), but requires
that you have a model of your data's energy spectre.
From: jer7@ns1.cc.lehigh.edu (JOEL ERIC RUDY)
Newsgroups: comp.dsp
Subject: Advice on DAS-1600 boards sought
Date: 6 Oct 92 19:16:18 GMT
Organization: Lehigh University

I am a college senior working with the DAS 1601 board from Keithley.  Does
anyone have any experience programming it?  If you do I would greatly
appreciate any advice, tips, pointers, etc. on how to program the board.
Thank you in advance.

Joel Rudy
Newsgroups: comp.dsp
From: croyer@bmers184.bnr.ca (Claude Royer)
Subject: References required on median filters.
Keywords: median filter, non-linear filter
Organization: bnr
Date: Thu, 01 Oct 92 19:33:53 GMT


I am looking for some reference litterature on the topic of
median and other type of non-linear filters. Anything to suggest?

Salutations,
Claude.

Newsgroups: comp.dsp
From: mlelstv@specklec.mpifr-bonn.mpg.de (Michael van Elst)
Subject: Re: FFT in 2D
Organization: Max-Planck-Institut f"ur Radioastronomie
Date: Mon, 28 Sep 1992 23:47:44 GMT

In <28SEP199214061890@ariel.lerc.nasa.gov> tohall@ariel.lerc.nasa.gov (Dave Hall (Sverdrup)) writes:
>    Here is the basic idea: Start with a 2-d array (matrix) of pixels. First,
>take the 1-d FFT of each row vector, forming an intermdiate matrix from the 
>result. Now take the 1-d FFT of each column vector in this matrix. The columns
>of the final output matrix are formed from these transforms. Therefore, the
>calculation of a 2-d FFT for an NxN matrix requires calculation of N^2 1-D
>FFT's.

Heh :-) Your explanation is right but of course it needs 2*N 1-D FFT's.

Regards,
-- 
Michael van Elst
UUCP:     universe!local-cluster!milky-way!sol!earth!uunet!unido!mpirbn!p554mve
Internet: p554mve@mpirbn.mpifr-bonn.mpg.de
                                "A potential Snark may lurk in every tree."
From: chuckc@hpnmdla.sr.hp.com (Chuck Corley)
Newsgroups: comp.dsp
Subject: Looking for DSP op system feedback
Date: 2 Oct 92 18:04:56 GMT
Organization: Hewlett Packard, Santa Rosa, CA


        We will be using a DSP processor to do a number of different
tasks.  Some of the tasks are DSP-related and some are general purpose
CPU tasks.  We are considering using a multi-tasking operating system
for the DSP processor.

        We've found two operating systems that look like good candidates:

1) The 'SPOX' operating system from Spectron Microsystems, and
2) The 'Nucleus' operating system from Accelerated Technology.

        I checked the recently posted FAQ file (thanks for posting this!),
but there was nothing there about DSP operating systems.

        Has anyone used or investigated either of these?  I know that
'SPOX' has a DSP math library where 'Nucleus' doesn't, but I'm not sure
how much of that math library I'll be using.

        Other than the math library and the price lists, I'm having
trouble seeing what the differences are between these two operating
systems.

        According to their brochure, the 'Nucleus' operating system is
also offered for a number of DSP and non-DSP processors.  Any comments
on it would be welcome, even if the comments were from experience with a
non-DSP processor.

	Any recommendations or comments you may have would be appreciated.

					Chuck Corley
					chuckc@sr.hp.com

The standard disclaimer:
The opinions and comments expressed here are mine and not those of my employer.
From: lazzaro@boom.CS.Berkeley.EDU (John Lazzaro)
Newsgroups: comp.dsp
Subject: Re: Real-time pitch shifting
Date: 1 Oct 1992 22:28:37 GMT
Organization: University of California at Berkeley
NNTP-Posting-Host: boom.cs.berkeley.edu

In article <1992Sep30.222908.18452@newshost.anu.edu.au> lonachon@mehta.anu.edu.au writes:
>
>Hi, I have just subscribed here as I am looking for algorithms or even some helpful
>pointers on real-time pitch shifting using DSPs.
>

Malcolm Slaney and I wrote a section in the comp.dsp FAQ posting addressing this question.


					



Newsgroups: comp.dsp
From: c8827148@cc.newcastle.edu.au
Subject: Wavelet Transform
Organization: University of Newcastle, AUSTRALIA
Date: Sun, 4 Oct 1992 03:37:37 GMT


I am currently studying the Discrete Wavelet Transform implemented in the
manner similar to the one described in Dr Dobbs Journal earlier this year. 
The FIR filters I am using are of my own design, giving much better 
reconstruction of the signal. 

My problem is that on a sampled sine wave input, I'm getting an
amplitude modulation effect on the reconstructed signal. The modulation
frequency itself seems to depend on the frequency of the input sine wave.

Has anybody else noticed this effect? What is the cause?

Thanks in advance,

Erik de Castro Lopo,
University of Newcastle,
Australia.

Newsgroups: comp.dsp
From: 4389750@mcimail.com  (TMS320 Hotline)
Subject: DSP-boards on VME
Nntp-Software: PC/TCP NNTP
Organization: Texas Instruments Materials and Controls Group
Date: Tue, 29 Sep 1992 20:56:38 GMT

In his message, <1992Sep23.113515.6355@fys.ruu.nl>, Jurriaan Vreeburg says,

>I'm startup up a project in which I want to use a modern (TMS320C30, 56001,
>96002, ADSP21xx) signal processor. I want to implement a system in a
>VME-environment. In this system I need some ADCs and a 68xxx-processor. Also,
>one or more DSPs must be present to perform some real-time, on-line processing
>of the ADC-data. So I'm looking for VME-boards which contain:
>- one or more DSPs
>- at least 256K word data memory
>- interrupt capability to VME
>- possibly some on-board ADCs & S/H (at least 14 bits at 1 Mhz)
>
>Can anyone help me in finding some manufacturors or distributors?

There are several companies who have developed VME boards for the TMS320C3x and
TMS320C4x DSPs. These companies include the following. For more information on
these and other TI or 3rd Party TMS320 Products you may contact your local
TI field sales office or call the TI Customer Response Center (CRC),
(214)995-6611, and request a copy of the TMS320 Development Support Reference
Guide and the TMS320 Third-Party Support Reference Guide.

ASR International                       03-502-5550 (Tokyo, Japan)
Atlanta Signal Processors, Inc.         (404)892-7265
Burr-Brown Corp.                        (602)746-1111
Catalina Research                       (719)531-5767
Image & Signal Processing, Inc.         (213)495-9533
Loughborough Sound Images Ltd.          (44)509 231843 (Loughborough, England)
Olsson Engineering                      (206)778-9480
Pacific Cyber/Metrix, Inc.              (415)829-8700
Pentek, Inc.                            (210)767-7100
Sky Computers, Inc.                     (508)250-1920
Spectrum Signal Processing Inc.         (604)421-5422
Star Technologies                       (800)782-7005

Regards,
TMS320 Hotline
(713)-274-2320       (voice)
(713)-274-2324       (fax)
(713)-274-2323       (BBS 300-9600,8,N,1)
4389750@mcimail.com  (Internet)


Xref: sparky comp.dsp:2296 comp.sys.ibm.pc.soundcard:2351
Newsgroups: comp.dsp,comp.sys.ibm.pc.soundcard
Subject: Re: realtime dsp with soundcards
From: deanaj@elec.canterbury.ac.nz (A. J. Dean)
Date: Sun, 4 Oct 1992 03:50:37 GMT
Organization: Electrical Engineering, University of Canterbury, New Zealand
X-Newsreader: TIN [version 1.1 PL6]

Nicolas Alejandro Arauz (na0s+@andrew.cmu.edu) wrote:
: Maybe someone can answer this question for me:
: > this, so I'll ask you.  My question that I wasn't sure about was whether
: > or not ANY of these cards would all simultaneous input/output from the
: > ADC/DAC do do digital effects?  Or perhaps even switch so fast that you

I've had a go a doing a similar thing with my soundcard (for guitar effects,
etc). BTW, it's a SB pro version 1. I think it may be able to do
simultaneous ADC and DAC (not in DMA mode) but even if it does the audio
routing/mixing circuitry won't. For a start, you can't turn off the input
monitor without turning down everything else including the DAC, ie you
always get some of your original signal straight through (though I did think
about subtracting it). Also, selecting the DAC disconnects any sound from
the ADC (well, it gives random values - could be used as part of the ADC).

You can switch rapidly between ADC and DAC, but the result is pretty awful,
though it did work. After that I gave up, along with giving up on my
ultra-optimised 8 bit FIR fiter routines for the PC. I'll wait/pay for a
soundcard which is a bit less limited!

Antony. (deanaj@elec.canterbury.ac.nz)
From: steve@lord.UUCP (Stephen F. Buchner)
Newsgroups: comp.dsp
Subject: Re: Hardware development tools for TMS320C40?
Summary: C40 Source debugger (Sun based)
Date: 29 Sep 92 21:50:57 GMT
Organization: Lord Corp., Cary, NC

In <1992Sep14.213228.6653@super.org> cfreese@super.org (Craig F. Reese) writes:

>I am searching for hardware (& associated software) development
>tools suitable for building small 'C40 based processing nodes.

>So far the only thing I have found seems to be the TI XDS510.
>This looks wonderful (and reasonably priced) but I was wondering
>if anyone knows of any third party setups.  Especially ones that
>DO NOT use a PC!

We have been using Swift, an X-windows based source level debugger from
Pentek.  They make VME and multibusII DSP and I/O cards with single/multiple
C25s, C30s and C40s.  I think the list on Swift was $2k.  We have been happy
with their C30 VMEbus cards, and the company is very responsive to user
requests/problems.  Our contact is Mario at (201)767-7100.  They have a good
selection of I/O and processor cards and daughter modules.  Drop a line if
you're interested.

Just a satisfied customer.
-Steve Buchner (steve@cx1gpx.lord.com)

Newsgroups: comp.dsp
From: 4389750@mcimail.com  (TMS320 Hotline)
Subject: TI DSP
Nntp-Software: PC/TCP NNTP
Organization: Texas Instruments Materials and Controls Group
Date: Tue, 29 Sep 1992 20:56:46 GMT


> Is there any public domain software available which implements the
> LPC-10 standard algorithm (2400bps)? I think the standard is
> FED-STD-1014.....not sure.

TI has LPC-10 available on its BBS for the C1x and C3x processors.  Here is
an overview on the algos:

	  Program	  Data		Processor
	  Memory	 Memory 	 Loading
	  ========	 ====== 	=========
C1x	   2K words	 2K words	   44%

C3x	   2.4K words	 2K words	    7%

--
Regards,
TMS320 Hotline
(713)-274-2320       (voice)
(713)-274-2324       (fax)
(713)-274-2323       (BBS 300-9600,8,N,1)
4389750@mcimail.com  (Internet)





JOYDEEP TRIPATHY
tripathy@saathi.ncst.ernet.in
620-1606 Ext.80

Newsgroups: comp.dsp,comp.sys.ibm.pc.soundcard
From: BrianW@SoundS.WA.com (Brian Willoughby)
Subject: Re: realtime dsp with soundcards
Reply-To: BrianW@SoundS.WA.com
Organization: SoundSoftware, Bellevue, WA, USA
Date: Wed, 30 Sep 1992 21:45:28 GMT

Nicolas Alejandro Arauz writes
| Maybe someone can answer this question for me:
| > I had been thinking about trying to write an effects program like you
| > did in your archive, only using the SoundBlaster card, or the PAS-16 or
| > the new Gravis card.  It seems as if you obviously know something about
| > this, so I'll ask you.  My question that I wasn't sure about was whether
| > or not ANY of these cards would all simultaneous input/output from the
| > ADC/DAC do do digital effects?  Or perhaps even switch so fast that you
| > wouldn't notice a lag.  Have you done any work with the commercial
| > soundcards?  I haven't gotten your archive yet but I will take a look at
| > it now.  It seems like you're possibly using high-sample rate DACs/ADCs
| > from reading your message.  Thanks f or your help!

While working on the MultiMedia Extensions to Windows 3.0 and Windows 3.1 at  
Microsoft, I worked closely with the SoundBlaster card (but not the other two  
that you mentioned).  The Creative Labs hardware cannot support simultaneous  
digital I/O for various reasons, one of which is that they only allocate one  
DMA channel in the PC for transferring the data.  Even at 8-bits mono, the PC  
has trouble with one direction, and cannot support A/D and D/A with th  
SoundBlaster.  This is true for SoundBlaster card versions 1.5, 2.0 2.5 and  
3.0.

Perhaps someone else has experience with the PAS-16 or Gravis card.
-- 
Brian Willoughby	Software Design Engineer, BSEE NCSU
BrianW@SoundS.WA.com	Sound Signal Processing Software and Consulting
NeXTmail welcome	- NO EMAIL SOLICITATION without prior permission
From: rkarlqu@hpscit.sc.hp.com (Rick Karlquist)
Newsgroups: comp.dsp
Subject: Star Semiconductor SPROC chip; opinions wanted
Date: 1 Oct 92 23:09:25 GMT
Organization: Hewlett-Packard, Santa Clara, CA

Does anyone have experience with the Star Semiconductor
"SPROC" DSP chip?  We saw a demo of it today and it looks
impressive, but there might be "gotcha's" that the salesman
glossed over.  Any success (or failure) stories out there?

Rick Karlquist
rkarlqu@scd.hp.com
From: bobb@vice.ICO.TEK.COM (Robert Beauchaine)
Newsgroups: comp.dsp
Subject: Re: Optimizing frequency resolution with DFT
Date: 1 Oct 92 23:05:54 GMT
Organization: Tektronix, Inc., Beaverton,  OR.

In article <1992Sep30.154718@sdec1.zrz.tu-berlin.de> feiten@sdec1.zrz.tu-berlin.de (Bernhard Feiten) writes:
>
>I'm looking for methods to analyze sounds with DFT.
>Normal window functions have the problem of either having low side lobe level
>or small main lobe bandwidth.

  The "problem of low sidelobe level and small main lobe bandwidth"?
  The lower your sidelobe level, the less spectral leakage problems
  you window will introduce.  The narrower you mainlobe, the better
  your spectral resolution.  The ideal window (an impulse) has no
  mainlobe width and no sidelobes whatsoever.  When creating a
  window, it can be shown mathematically that you will *always*
  trade mainlobe width for reduced sidelobe level, at least in all
  physically realizable windows.

>
>Does anybody knows an algorithm based on DFT (FFT) that optimizes both - 
>for example a main lobe bandwidth similar to a rectangular window and 
>side lobe level similar to a Kaiser-Bessel window ? 

  I refer you to the definitive window reference, 

  "On the Use of Windows for Harmonic Analysis with the Discrete
  Fourier Transform", Frederic J. Harris, Proceedings of the IEEE,
  Vol. 66, No. 1, January, 1978.

  Additionally, for a reference applicable to sidelobe behavior:

  "Some Windows with Very Good Sidelobe Behavior", Albert H.
  Nuttall, IEEE Transactions on A.S.S.P., Vol ASSP-29, No. 1, Feb.
  1981.

  Another good reference for a fascinating window is:

  "A Novel Window for Harmonic Analysis", Fung I. Tseng, Tapan K.
  Sarkar, and Donald D. Weiner, IEEE Transactions on A.S.S.P., Vol
  ASSP-29, No 2, April 1981.

  The latter allows the user the ability to create a very deep notch
  in the sidelobe level at a selectable frequency displacement from
  the main lobe.  Good for harmonic/multitone signals.

/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\

Bob Beauchaine
bobb@vice.ico.tek.com
Newsgroups: comp.dsp
From: ciminero@ucunix.san.uc.edu (Ronald N. Ciminero)
Subject: Customizing DFDP3/plus for use with TI's EVM
Keywords: DFDP3/plus, TI's EVM, TMS320
Organization: University of Cincinnati
Distribution: usa
Date: Tue, 6 Oct 1992 14:45:54 GMT
Expires: Tue, 20 Oct 1992 04:00:00 GMT

What steps must be taken to customize Atlanta Signal Processor's
DFDP3/plus software package for use with TI's EVM development hardware
board using the TMS320 processor?

I am writing this request for a colleague who does not have access to
network news but does have an e-mail address. Since I am not a regular
reader of this group, please send your responses and questions
to Dennis Farrell at:

	farrell@caseet.cas.uc.edu

Thanks!
Ron
-- 
--------------------------------------------------------------------
R.N. Ciminero			Bitnet:   ciminero@ucbeh
University of Cincinnati	Internet: ciminero@ucunix.san.uc.edu
College of Applied Science
Newsgroups: comp.dsp
From: dak@messua.informatik.rwth-aachen.de (David Kastrup)
Subject: Re: Optimizing frequency resolution with DFT
Organization: Rechnerbetrieb Informatik  /  RWTH Aachen
Date:  7 Oct 92 00:34:43 GMT

sjreeves@eng.auburn.edu (Stan Reeves) writes:


>In order to optimize the tradeoff between main lobe width and side lobe
>height, one has to have a criterion.  Do you want more resolution and more
>variance or do you want less variance and less resolution?  I suppose that
>one could come up with a criterion based on expected mean-square error
>between the estimated and actual power spectrum, but I don't know of any
>neat closed-form expressions for this.

Well, if you define bandwidth (and timespread) as the variance of a
similar-spread probability density function, you get a sort of
square mean error. The product of the time width and the bandwidth
is limited downwards, it cannot get better than an optimum, that is
provided by a gaussian curve which again transforms into a Gaussian
in the frequency domain. You cannot get any better, but this is definitely
a closed-form expression. So mean square criterions will not get you
too far, but will show you the direction.
From: batavia@phakt.usc.edu (Parag Batavia)
Newsgroups: comp.dsp
Subject: 'C40 design examples....
Date: 28 Sep 1992 18:12:06 -0700
Organization: University of Southern California, Los Angeles, CA
Distribution: usa
NNTP-Posting-Host: phakt.usc.edu

Hi...Does anyone here have any references to design examples using the 'C40.
Anything dealing with building a minimalist system would be helpful...Also,
if anyone knows of any techincal articles dealing with topics such as 
benchmarks and examples of parallel use, that would be good too...Thanks much..

Parag Batavia
USC Robotics Lab

parag@mensa.usc.edu
batavia@usc.edu
From: mshiels@TMSoftware.Ca (Michael A. Shiels)
Newsgroups: comp.dsp
Subject: ADPCM documentation, and other compressions
Date: Tue, 29 Sep 1992 00:38:55 GMT
Organization: MaS Network Software and Consulting


Where can I find documentation on the various voice compression schemes?
I really need ADPCM to decompress it for reading with Windows 3.1 WAV
-- 
---

Michael A. Shiels                           | mshiels@masnet.uucp
MaS Network Software and Consulting         | mshiels@tmsoftware.ca
Xref: sparky comp.dsp:2243 sci.math.num-analysis:2850