HDTenEightyP/NetNews huggingface dataset:
From: gp310ad@prism.gatech.EDU (ROBERT DUCKWORTH)
Newsgroups: comp.graphics
Subject: Dupont 4Cast for sale. 11x17 dye sub postscript printer. like new.
Keywords: CYMK C color postscript pre-press photo
Date: 2 Sep 92 12:35:28 GMT
Distribution: na
Organization: Georgia Institute of Technology

We have a Dupont 4Cast printer we would like to sell.
This is a 300DPI, dye sublimation, postscript, 11x17, full bleed, CYMK 
printer.
Less than 500 prints.
All original docs and packing.
We'll include a new supplies kit.

Why are we selling?
While this is an excellent CYMK printer, we are primarily interested in
RGB (photographic) printing. The artist/photographers are not happy with
any of the conversion algorithms we have come up with.

So, if you're happy with very accurate Pantone color reproduction or
have another use for a great CYMK printer, this is a deal.

It is set up to work with a Mac.

$15000.00

(Current new price with discounts is over $30000.00)

404-874-5051
404-892-2301 (fax)

Also have a 4D25TG to dispose of.
340meg, 150meg tape, 12meg RAM, 24bit, 24bit Z, turbo graphcs installed.
19" SGI color tube. It's in great shape for a 3 year old machine. $7250.00

-bob

Anyone got a good/fast RGB to CYMK algorithm/program :-)
?????

-- 
Bob Duckworth  (WB4MNF)         960 Ralph McGill Blvd. Atlanta GA 30306-4447
uucp: ...!{allegra,amdmhplabs,ut-ngp}!gatech!prism!gp310ad       -rave on-
Internet: gp310ad@prism.gatech.edu  404 - 874-5051(H) 888-0389(O) 892-2301(F)
Newsgroups: comp.graphics,comp.windows.x
From: jack@shograf.com (Jack Ritter)
Subject: PEX source programs
Organization: SHOgraphics
Date: Wed, 02 Sep 1992 16:59:44 GMT
Followup-To: poster
Keywords: PEX X windows

We are looking for PEX programs to test on
our PEX server. We have the big apps, like AVS,
but would like anything written in PEX that
is debugged. In return, you see what happens
when your code runs on another server, which
is a very good way to find new bugs (our server
bugs in particular, but maybe even a few of yours!).

----------------------------------------------------
   If only I had been compiled with the '-g' option.

          Jack Ritter. SHO Graphics.
          1890 N Shoreline Blvd.
          Mountain View, CA 94043
          email: jack@shograf.com
          Tele:  (415) 903-3867
From: dmp@eeyore.cs.bham.ac.uk (Donald Peterson)
Newsgroups: comp.cog-eng
Subject: FINAL CALL : AISB'93
Date: 3 Sep 92 15:17:31 GMT
Organization: Birmingham University Computer Science


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

                          AISB'93 CONFERENCE

                        FINAL CALL FOR PAPERS 

    Theme: "Prospects for AI as the General Science of Intelligence"

                       29 March --  2 April 1993 

                        University of Birmingham

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

The Society for the Study of Artificial Intelligence and the
Simulation of Behaviour will hold its ninth bi-annual conference on
the dates above at the University of Birmingham.

This is a reminder that the deadline for submitted papers and posters
is 15 September 1992 (though a few days leeway will be allowed). 

Papers are invited in any of the normal areas represented at AI and
Cognitive Science conferences, including:

AI in Design,
AI in software engineering
Teaching AI and Cognitive Science,
Analogical and other forms of Reasoning
Applications of AI,
Automated discovery,
Control of actions,
Creativity,
Distributed intelligence,
Expert Systems,
Intelligent interfaces
Intelligent tutoring systems,
Knowledge representation,
Learning,
Methodology,
Modelling affective processes,
Music,
Natural language,
Naive physics,
Philosophical foundations,
Planning,
Problem Solving,
Robotics,
Tools for AI,
Vision,

The conference proceedings will be published. Long papers and invited
papers will definitely be included. Selected poster summaries may be
included if there is space.


                 PAPER LENGTH and SUBMISSION DETAILS

Full papers:
    10 pages maximum, A4 or 8.5"x11", no smaller than 12 point print
    size Times Roman or similar preferred, in letter quality print.

Poster submissions
    5 pages summary

Excessively long papers will be rejected without being reviewed.

All submissions should include

    1. Full names and addresses of all authors
    2. Electronic mail address if available
    3. Topic area
    4. Label: "Long paper" or "Poster summary"
    5. Abstract no longer than 10 lines.
    6. Statement certifying that the paper is not being
       submitted elsewhere for publication.
    7. An undertaking that if the paper is accepted at least
       one of the authors will attend the conference.

THREE copies are required.


                  EMAIL, PAPER MAIL, PHONE and FAX. 

Email:
    * aisb93-prog@cs.bham.ac.uk 
      (for communications relating to submission of papers to the programme)
    * aisb93-delegates@cs.bham.ac.uk 
      (for information on accommodation, meals, programme etc. as it 
       becomes available --- enquirers will be placed on a mailing list)

Address:
    AISB'93 (prog) or AISB'93 (delegates), 
    School of Computer Science,
    The University of Birmingham,
    Edgbaston,
    Birmingham,
    B15 2TT, 
    U.K. 

Phone:
    +44-(0)21-414-3711

Fax:
    +44-(0)21-414-4281

----------------------------------------------------------------------------
Xref: sparky comp.research.japan:227 comp.ai:3231
From: thomas@smaug.uio.no (Thomas Gramstad)
Newsgroups: comp.research.japan
Subject: VR in Japan?
Date: 29 Aug 92 21:04:49 GMT
Reply-To: simstim@smaug.uio.no
Followup-To: poster
Organization: University of Oslo


I would like to get in contact with people involved in VR
research and projects in Japan.  One reason for this is to
invite Japanese to our VR conference in Norway next year
(see conference posting in sci.virtual-worlds).  Please
respond directly to me (I will summarize back to the net,
if interest) at the address simstim@smaug.uio.no.  Thanks.

                    Thomas Gramstad
From: roth@sitter.ips.id.ethz.ch (Martin Roth)
Newsgroups: comp.research.japan
Subject: Optical Music Recognition
Summary: Any pointers to recognition of music?
Keywords: pattern recognition, omr, ocr, notation
Date: 2 Sep 92 13:16:19 GMT
Organization: ETH Zurich (Swiss Federal Institute of Technology)


I'm working on a program for recognition of printed music.
Now I'm looking for pointers to literature and people concerning
the same subject. Any infos appreciated, email to
roth@ips.id.ethz.ch.

So far I have contacts to McGill University, Canada and University of
Surrey, England.

I read about the WABOT Keyboard Playing Robot of Waseda University,
Tokyo. Anyone knows if someone since worked on the recognition of
music ?

_______________________________________________________________________
      _   Martin Roth                Martin Roth    ETHZ, ips, RZ F16
|\  /|_)  Mail: roth@ips.id.ethz.ch  Sandacker 14   g 01/256 55 68
| \/ | \                          CH-8154 Oberglatt p 01/850 32 75
          [Ing. Comp. Sci.]          Switzerland    emails welcome!
-----------------------------------------------------------------------
[Attention: from-field is wrong, use send to roth@ips.id.ethz.ch]

Xref: sparky comp.research.japan:223 comp.graphics:9179
From: kace@enel.ucalgary.ca (Ray Kacelenga)
Newsgroups: comp.dsp
Subject: Factored State Variable Description of Digital Filters (FSVD)
Date: 25 Aug 92 18:32:57 GMT
Organization: ECE Department, U. of Calgary, Calgary, Alberta, Canada

In the book `Digital Signal Processing', R.A. Roberts and C.T. Mullis have
presented an algorithm for calculating the output roundoff noise of digital
filters using a factored form of the signal flow graph. The algorithm 
essentially does the following:
	
	1. obtain primitive SFG which gives the variable classes
	2. obtain transition matrices
	3. merge the last matrix G0 with its neighbour
	4. compute the K and W matrices using the ABCD matrices
	5. compute the Ki's and Wi's corresponding to the transition matrices
	6. compute the noise gain contribution from each factor as the sum of 
	   the Ki*Wi for each non trivial row
	7. sum the noise contributions from all the factors to get the total
	   noise gain.

The trouble is that the corresponding K and W matrices are not the same
dimensions unless shifted by one relative to each other. The other problem
is that in D.M. Henrot's Ph.D. dissertation, a similar algorithm is presented.
The algorithm goes through steps 1 and 2 above. In step 3, G0 is not merged.
Step 4 is the same. The calculation of the Ki's is the same. The calculation
procedure of the Wi's is different; in particular the initialization is not the
same. Henrot scales the factors and computes the total noise gain as the sum of
all the Wi traces. The two algorithms give very different results.

I am looking to reconcile the two algorithms or to figure out the correct one
to use. It seems to me that Henrot's algorithm would be correct if he picked
out, from the factors, only those rows with non-trivial multiplier 
coefficients.

I am looking to hear from anyone who has had occassion to use either one of the
algorithms and in what perspective they view the whole process of computing the
noise in this manner.

Please reply using the e-mail address below:

		kace@enel.ucalgary.ca

Ray Kacelenga (Ph.D. candidate).
Newsgroups: comp.dsp
From: z98t@jupiter.sun.csd.unb.ca (Vijay Parsa)
Subject: Re: NEED WAVELET APPLICATIONS INFO. FOR...
Organization: University of New Brunswick
Date: Fri, 28 Aug 1992 15:26:44 GMT

In article <5wqy3l_@rpi.edu> sassoj@aix.rpi.edu (John J. Sasso Jr.) writes:
>  Does anyone know of any articles on the applications of wavelets to sound,
>such as audio compression, etc.?  Also, any articles on the applications of
>wavelets to biomedical engineering, such as EKG, EEG, ECG, etc.?  Any info
>is appreciated.
>
>                     John Sasso
I'd like to have the same info please (esp the biomedical part). Thanks in
advance.
vijay.


Newsgroups: comp.dsp
From: u9066624@cc.uow.edu.au (Tippu Hassan)
Subject: TMS320C25: Continuous speech problem
Organization: University Of Wollongong
Date: Wed, 2 Sep 92 05:54:08 GMT

Hello anyone,

My name is Tippu and I live in Australia. I am working with the TI
"SWDS" (Software Development Systems) board which is a support tool
for the TMS320C25 dsp chip. I am using the TMS320C25 serial port to
read in digitized speech samples and then storing them on external
RAM memory which is on the SWDS board. The SWDS only has 32K bytes
of onboard RAM avialable and therefore I can only store 2 seconds
worth of 16-bit digitized speech. So in order to record continous
speech I am using the cpu processor to upload these speech samples
from the SWDS board RAM and store them on files once the RAM is full. 
This is done via C applications written for the cpu. 
My problem is that in order for the cpu to access the SWDS RAM once 
it is full, it MUST place the TMS320C25 in halt mode, then upload 
the speech samples and then unhalt the TMS320C25 to resume reading 
in the next load of speech samples. But the cpu cannot execute this 
process quite enough, and TMS320C25 loses some speech samples while 
it is halted. I cannot uses the grep command on the TMS320C25, which
makes the external RAM global, because it requires external circuitry
for the TMS320C25 and it is wired onto the SWDS board. Can anyone offer
any ideas or help as it would be much appreciated.
From: jyrkkari@sara.cc.utu.fi
Newsgroups: comp.dsp
Subject: DSP and filter design
Date: 31 Aug 92 09:05:19 GMT
Organization: University of Turku, Finland


   	
A few questions about DSP:
 -Are there any (puplic domain or shareware) programs for digital filter
   design for PC ? Where can I get them ? 
 -Is there any anonymous FTP sites where there are DSP routines coded with c ?

     Kari Jyrkkala
From: jbuck@forney.berkeley.edu (Joe Buck)
Newsgroups: comp.dsp
Subject: Re: Hilbert Transform (?)
Date: 3 Sep 1992 19:25:20 GMT
Organization: U. C. Berkeley
NNTP-Posting-Host: forney.berkeley.edu

In article <LcVHqB1w165w@precipice.chi.il.us> jjw@precipice.chi.il.us (John Welch) writes:
>    I would like to make a DSP project that would digitize
>voice grade audio (300-3000 Hz) and provide the original signal
>and that signal phase-shifted by 90 degrees.  I believe this 90
>degree phase shift is done with a Hilbert Transform.  I have found
>the name, but no info on *how* to do a Hilbert Transform.  Any
>assistance would be greatly appreciated.  -->jjw

A Hilbert transform is simply a filter whose frequency response is j
(magnitude 1, phase 90 degrees) for positive frequencies and -j (magnitude
1, phase -90 degrees) for negative frequencies.  The exact Hilbert
transform is unrealizable, just as the perfect low-pass filter is
unrealizable; however, you only need this response in the 300-3000 Hz
bandwidth.  One good way to do this is to use the Parks-McClellan
algorithm to design an FIR filter that approximates the Hilbert transform
in this band (with arbitrary response elsewhere).  I have an ancient
Fortrash program that can do this; it is included as part of the Ptolemy
distribution for those that have that.  Another way that's not as good is
to simply truncate the ideal Hilbert transform impulse response to finite
length and add a delay.  The ideal impulse response is

h(n) = 2/(n*pi) for n odd
0               for n even

and that's for both positive and negative n.







--
Joe Buck	jbuck@ohm.berkeley.edu
Newsgroups: comp.dsp,comp.compression
From: enis@flower..bellcore.com (A. Enis Cetin)
Subject: LPC10e document
Summary: vocoder 
Keywords: DSP, voice coder
Reply-To: enis@flower.UUCP (xxxx)
Organization: Bellcore MRE
Date: Thu, 3 Sep 92 20:20:40 GMT

I am looking for 

D.P.Kemp, et. al., "LPC-10e Enhanced LPC-10 Voice coder, 
Phase 1 Report,"
National Security Agency, document number R163-84-01", 
March 4, 1985.

Is there a site where this document can be obtained
via ftp.

Thanks in advance.
Enis
Newsgroups: comp.dsp,comp.sys.next.programmer
From: phaedrus@leland.Stanford.EDU (Avery Wang)
Subject: Fast arccos for the DSP 56001 code here
Organization: DSG, Stanford University, CA 94305 USA
Date: Mon, 31 Aug 92 15:56:59 GMT

Hi DSP hackers-
A few months ago I wrote a simple inverse-table lookup algorithm for an inverse
cosine on the DSP 56001 from Motorola.  The output is normalized to between 0
and 127 = (0 to Pi).  You get 7 bits of accuracy, which is all you can really
get from an inverse Sine-ROM lookup.  Incidentally, this algorithm can be
easily adapted to do an inverse table lookup on any monotonically decreasing
function in tabular form (128 elements).

All I ask is for you to send me email telling me that you think it's useful, so 
I'll feel good  and be encouraged to put more stuff on the net in the future.

=======================================================================
;   arccos.asm
;   Calculates the arc-cosine of the input to 8 bits.
;   The output is between 0 and +127 and is in a
;
;   Copyright! February 20,1992
;   Free for non-profit usage
;
;   Avery Wang                         
;   Ruhr-Uni-Bochum
;   Institut fuer Neuroinformatik
;   ND03, Postfach 102148
;   W-4630 Bochum
;   Germany
;   avery@neuroinformatik.ruhr-uni-bochum.de  or
;   awang@isl.stanford.edu

; uses the Sine-ROM and simply does an inverse table lookup.
; input and output are in a
; uses a,b,x0,y0,r6,r7,n7
arccos macro
    move #$140,r7           ;offset
    move r7,r6
    move #$40,n7            ;guess
    move n7,b
    do #7,_arccos_loop
        move (r6)+n7                  ;update guess
        asr b       a,x0    y:(r6),y0 ;make increment smaller, lookup cosine
        cmp y0,a    b,n7
        tlt x0,a    r6,r7       ;dummy move, if arg>table value, keep old guess
        nop
_arccos_loop
    move r7,a
    move #$140,y0
    sub  y0,a                           ;result is here

    endm

Newsgroups: comp.dsp
From: DSPWorld@world.std.com (Amnon Aliphas)
Subject: ICSPAT & DSPWorld UPDATE
Keywords: DSP
Organization: The World Public Access UNIX, Brookline, MA
Date: Fri, 28 Aug 1992 19:45:31 GMT

The program for ICSPAT & DSP World Expo. cannow be obtained:

contact:    DSP Associates
            18 Peregrine Road
	    Newton Centre, MA
	    U.S.A.

	    Tel: (617) 964-3817
	    Fax: (617) 969-6689

	    email      DSPWorld@world.std.com


	The International Conferece on Signal Processing

	             Applications & Technology

			    featuring

			  DSP World Expo.

    Hyatt Regency in Cambridge, Massachusetts - November 2nd - 5th, 1992



The conference and Expo. will feature over 200 applications by design engineers
and researchers from over 35 countries worldwide. The International Conference
on Signal Processing Applications and Technology focuses strictly on industrial
applications and product development based on signal processing technology,
attracting a large number of engineers representing domestic and international
corporations.

Some of the companies represented at DSP World Expo. are;


	Analog Devices	  		Ariel Corporation
	AT&T Microelectronics		CADIS GmbH
	Catalina Research		Comdisco Systems Inc.	
	CSPI				Data Translation
	DSP Research			DSP Software
	Dynetics			GEC-Plessey Semiconductor
	hema Electronik GmbH		Heurikon
	Hyperception			IEEE Spectrum Magazine
	Image & Signal Processing	Intelligent Systems Int'l.
	Ixthos				Loughborough Sound Images
	Mercury Computers		Momentum Data Systems
	Motorola			National Instruments
	Sharp Microelectronics		Signal Proc. Assoc. Pty.
	Sonitech International		Spectrum Signal Processing
	Star Semiconductor		Sunnyside Inc.
	Texas Instruments		The Math Works
	Zoran.				Transtech


If you would like to reserve exhibiting space 

	ICSPAT / DSPWorld Expo.

	DSP Associates			Tel: (617) 964-381718
	Peregrine Road			Fax: (617) 969-6689
	Newton Centre, MA 02159, USA	e_mail address DSPWorld@world.std.com
From: freed@europa.orion.adobe.com (Alex Freed)
Newsgroups: comp.dsp
Subject: Speech recognition?
Date: 31 Aug 92 20:26:21 GMT
Organization: Adobe Systems Inc


Hi,

I wonder if there is any code or algorithm that could be implemented on an
inexpensive DSP like say TMS32015 to recognize a few simple words like numbers
from 0 to 9 and letters. Not nesessarily speaker independant. Any leads?
--
 _______________________________________________________
| -Alex Freed (The opinions expressed are my own.   	|		    
|	        However everyone is entitled to them.)	|		    
| freed%adobe.com@uucp-gw-1.pa.dec.com			|
 -------------------------------------------------------
Newsgroups: comp.dsp
From: morrow@cns.ucalgary.ca (Bill Morrow)
Subject: Bugs in 68HC16 project
Date: Wed, 26 Aug 92 16:10:18 GMT
Organization: Dept. Clinical Neurosciences, U of Calgary, Alberta, Canada

He is having problems with the IIR filter section. I seem to remember
some comments about there being typos in the instruction manual, or
bugs in the code. If so, could someone mail the corrections to me?

Thanks,
Bill

--
__________________________________________________________
Bill Morrow                    voice: (403) 220-6275
Clinical Neurosciences         fax: (403) 283-8770
University of Calgary          e-mail: morrow@cns.ucalgary.ca
From: 5140840@mcimail.com
Newsgroups: comp.dsp
Subject: TI DSP
Date: 4 Sep 92 15:00:19 GMT
Organization: Texas Instruments Materials and Controls Group
Nntp-Software: PC/TCP NNTP

>  Has anybody heard of a C40 application card with an A/D
>  and possibly a D/A.

One A/D - D/A card on the market is a card from Data Translation (the
DT3801) that has six types of data acquisition - one on each C40 commport. The
card is really meant for OEM users for automatic test equipmenet, inspection,
etc., but it could be used as an application board (it even runs with SPOX).

Another possibility is to try DSP Research (408-773-1041). While their "Tiger
40" board does not have A/D and D/A directly on-board, it does have A/D
daughtercards that can connect to the board.  ASPI (404-892-7265) also has
A/D cards for its C40 "Vortex" board.

Most the other C40 third parties will be coming out with A/D cards as well
over the next few months.

Regards,
Mitch Reifel
Texas Intruments
Houston, TX
Newsgroups: comp.dsp
From: williamb@ee.ubc.ca (william burchill)
Subject: Re: Sliding FFTs
Organization: University of BC, Electrical Engineering
Date: Wed, 26 Aug 1992 06:21:13 GMT

Sliding FFT's are useful for finding the varying frequency content 
of a signal as a function of time.  If the frequency content changes
slowly it is sufficient to slide several samples between FFT evaluations.
This is a hopping FFT.  A decade ago I use a hopping FFT as part of a 
speech spectrograph system.  The familiar 3D pictures of speech signal's
frequency content which look like mountain ranges are a typical application.
The longer the hop the less calculation needed, but time resolution is 
reduced.
 

Newsgroups: comp.dsp,news.answers
From: heifetz!rphroy!re3sc139h.gmr.com!wjs@uunet.UU.NET
Subject: REQ: Info re: third partys for C40 chip
Followup-To: comp.dsp
Organization: Electronic Data Systems - Test Engineering Systems
Date: 1 Sep 92 14:46:05 GMT


I apologize if this information has been posted recently.

I am looking for a list of third party vendors of the TMS320C40
based processor boards.  I would be particularly interested in
VME based multi-processor configurations.

Any information available on good C, C++, or Fortran compilers
would also be appreciated.


							   Thanks in advance,

							   W. Stuart   wjs@re3sc139h.gmr.com
From: frandsen@seismo.CSS.GOV (Marvin Vaun Frandsen)
Newsgroups: comp.dsp
Subject: C Radix bidirectional FFT
Date: 26 Aug 92 00:53:57 GMT
Distribution: all
Organization: Center for Seismic Studies, Arlington, VA

Desparately seeking radix FFT:  I am looking for a split radix
FFT that will perform FFT and inverse FFT.  Preferred language is
C.

The archie files list tons of FFTs, but it would take a long time to
ftp and review them.  Recommendations for good choices fulfilling the
above would be very appreciated.  (Good also includes fast, well
written and easy to use, but just working would be a good start.)

Thanks very much in advance for the help.

-- Marv Frandsen     frandsen@beno.css.gov

-- 
-------------------------------------------------------------------
	  Marvin Vaun Frandsen       frandsen@beno.css.gov
	    
Newsgroups: comp.dsp
From: smb3u@squeeg.psyc.virginia.edu (Steven M. Boker)
Subject: Re: Sliding FFTs
Organization: University of Virginia, Department of Psychology
Date: Tue, 1 Sep 1992 19:49:45 GMT

I used the technique that William so succinctly described to track
blood flow in the uterine artery from ultrasound data in a study
at the University of Colorado.  Since the reflection of ultrasound
from a moving target produces a doppler effect, the spectrum of
frequencies recorded from the reflection of an artery gives a good
idea of the differing velocities of blood flow in the artery.

Arterial constriction and dyastolic and systolic blood flow could be
quite easily seen in the resulting sonogram.  I released the code
to the public domain.  You can find it in the edsnd package for the
NeXT at princeton.edu and sonata.cc.purdue.edu.

Steve

-- 
 #====#====#====#====#====#====#====#====#====#====#====#====#====#====#
 #  Steven M. Boker            #          "Two's bifurcation           #
 #  smb3u@virginia.edu         #           but three's chaotic"        #
 #====#====#====#====#====#====#====#====#====#====#====#====#====#====#