HDTenEightyP/NetNews huggingface dataset:
From: f88mattb@fy.chalmers.se (Mattias Bryntesson)
Newsgroups: comp.dsp,sci.math.num-analysis
Subject: Re: Delayed signals
Date: 28 Sep 92 11:36:05 GMT
Distribution: comp
Organization: Chalmers University of Technology, G|teborg, Sweden

In article <SINKKONE.92Sep25152003@klaava.Helsinki.FI> sinkkone@klaava.Helsinki.FI (Janne Sinkkonen) writes:
>
>Our goal is to estimate the values of a smooth, evenly sampled
>function x(t) between samples, i.e. at points x(t+r), where r is a
>real number (let's assume sampling was done at points t=0,1,2,..,N-1).
>

Sounds like an interpolation, and should be obtained by convolution with
sinc(x) := (sin (pi*x) )  / (pi*x). With this notation

   x(t+r) =            sum           [ x(k)*sinc(k+r) ]
            k= ..., t-1, t, t+1, ...

where you should sum over all samples (=enough samples around your point).

I've got a book by Bracewell that covers this and a lot of other
Fourier stuff. I think it's called "The Fourier Transform and its
Applications".

Mattias
Newsgroups: comp.dsp,comp.sys.ibm.pc.soundcard
From: dedmunds@selkirk.sfu.ca (Darran Edmundson)
Subject: Re: Gravis Ultrasound
Organization: Simon Fraser University, Burnaby, B.C., Canada
Date: Fri, 2 Oct 1992 22:07:05 GMT

adhir@cygnus.umd.edu (Alok Dhir) writes:

>So... Did it really come out this time?  Is it in anybody's machine, singing
>away happily?  What's the scoop?!

Not yet.  Apparently they are still shrink wrapping in the factory (according
to the salesman Chris and the owner of Software Alley here in Vancouver).  My
name is 2nd on the list at SA which will likely have them late today or
early tommorrow.  I wouldn't expect delivery before early next week if you
live outside of the Vancouver area (just a guess).

-- 

Darran Edmundson
darran@chaos.phys.sfu.ca

Newsgroups: comp.dsp
From: Bob.Smith@bbs.oit.unc.edu (Bob Smith)
Subject: Experiences with Microway "Number Smasher 860" ???
Organization: Extended Bulletin Board Service
Date: Thu, 1 Oct 1992 22:06:13 GMT

Hello,

I've not yet heard a peep about Microway or their Intel I860 based
vector processor board for ISA/EISA PCs.  Anyone out there have any
first hand experiences with this board?

Also, anyone using Microway's NDP C and Fortran compilers?

thanks in advance,
bob smith

--
   The opinions expressed are not necessarily those of the University of
     North Carolina at Chapel Hill, the Campus Office for Information
        Technology, or the Experimental Bulletin Board Service.
           internet:  bbs.oit.unc.edu or 152.2.22.80
Newsgroups: comp.dsp
From: knud@enel.ucalgary.ca (Steve Knudsen)
Subject: Advice on Mac DSP boards sought
Date: Mon, 05 Oct 92 06:48:45 GMT
Distribution: na
Organization: ECE Department, U. of Calgary, Calgary, Alberta, Canada
Keywords: DSP boards, image processing, fast transforms, DSP research

Hello,

I'm interested in getting a DSP board or two to put into a Quadra
950. I will use the boards for multidimensional signal processing
research; mostly image and image sequence processing.

I would like to have any opinions regarding:
  - boards and chips available
  - software and hardware support for said boards
  - personal preferences of users
  - educational uses (as I might use the hardware in a course
    I'll give)
  - etc.

Of course, if there is interest, I'll summarize my replies.
Please e-mail me the answers

Steven Knudsen
(for now) knud@enel.ucalgary.ca




-- 
K. Steven Knudsen                     | The lasers are in the labs, the old
University of Calgary                 | man is dressed in white clothes.
Department of Electrical Engineering  | Everybody says that he's mad - 
knud@enel.ucalgary.ca                 | Nobody knows the things that he knows.
Newsgroups: comp.dsp
From: thinman@netcom.com (Technically Sweet)
Subject: Re: ADPCM documentation, and other compressions
Organization: International Foundation for Internal Freedom
Date: Fri, 2 Oct 1992 04:27:16 GMT

mshiels@TMSoftware.Ca (Michael A. Shiels) writes:


>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

FTP to ftp.cwi.nl and get pub/adpcm.shar.  It is one ADPCM implementation.
Might help.

-- 

Lance Norskog

Data is not information is not knowledge is not wisdom.
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: Fri, 2 Oct 1992 13:33:06 GMT

> 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.

TI will be publishing a C40 application book in 1Q93 that has design examples,
code, etc., but it is not ready yet at this time.

In the meantime, here are some places to find some info that may be helpful:
  o Design examples - TI publishes the schematics of its C40 "PPDS" development
    system and even puts the Orcad model for it on the TI BBS.  While this is
    not a minimalist system (it has four processors), it may help some.
  o Software/parallel use - TI has published two ap notes for the C40 (which
    will be "rolled" into the ap book).  The first is a parallel matrix
    multiplication ap note and the second is a parallel 2-D FFT ap note.  The
    literature numbers for these are SPRA026 and SPRA027, respectively.  Also,
    non-parallel code (FFTs, filters, etc.) can be found in the C40 user's
    guide (lit number SPRU063).  Note that all code can also be found on the
    BBS.  This literature is available through TI distributors, TI sales
    offices, and our Customer Response Center (800)-336-5236, and the TM320
    Hotline.

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


Newsgroups: comp.dsp,comp.sys.ibm.pc.soundcard
From: jokim@jarthur.claremont.edu (John H. Kim)
Subject: Re: Music/sound program compatibility under OS/2? Anyone having probs?
Organization: Harvey Mudd College, Claremont, CA 91711
Date: Fri, 2 Oct 1992 06:52:45 GMT

feoh@Gnu.ai.mit.edu (Chris Patti) writes:

>hello, I'm buying a 486DX50 in 3 weeks, I'm currently trying to decide whether
>or not to go with MSDOS/Windows (I despise DOS, but want full compatibility
>with most windows apps) or OS/2 2.0 (a MUCH BETTER OS for *MY* needs, no flames
>:)..

>Could anyone who's having problems running sound/music apps under OS/2 let me 
>know/post it here? It'd be interesting to find out if IBM's claims are true :)

Well, I'm on a "lowly" 386/33 32k cache 8MB ISA 160 Meg 16ms ESDI drive
with a SBPro.  (Whew, what a mouthful)

The only problem I've had with games using the SB is the digitized stuff
on Falcon 3.0.  A couple other games (Wing Commander, Wolfenstein 3D)
using digitized stuff works OK.  I've had no problems with music other
than pauses or slowdowns during heavy disk activity.  Playing MODs with
WOWII under OS/2 gave some barely noticeable (but nonetheless irritating)
pauses between each portion of music sent through the buffer.  The MIDI
program and all the other neat stuff that comes with the SBPro works
just fine (except for the slowdown mentioned above).

I would say with a 486DX50, if you get at least 16 MB (to cut down on
swapping and disk activity) and possibly EISA and/or SCSI-2, you should
be safe 98% of the time.  You could use boot manager to boot a DOS partition
just in case you ever run into problems.
-- 
John H. Kim                 | (This space to be filled when I
jokim@jarthur.claremont.edu | think of something very clever 
uunet!jarthur!jokim         | to use as a disclaimer)
Newsgroups: comp.dsp
From: 4389750@mcimail.com  (TMS320 Hotline)
Subject: 2D FFTs
Nntp-Software: PC/TCP NNTP
Organization: Texas Instruments Materials and Controls Group
Date: Tue, 29 Sep 1992 20:56:27 GMT

In his message, <1992Sep22.163256.15935@versyss.com>, Keith Doyle says

>I'm interested in using FFT for noise reduction of images, and while I
>have worked with FFT in 1D (for audio) and have an assortment of FFT
>source code for 1D, I don't really know how to apply it to the 2D case.
>Is it simply an FFT of all the rows combined with an FFT of all the columns
>of pixels in the image?  Or is it more complex than that?  Is there a
>reference and/or PD source code someone can point me to?

There is a reference and source code. The reference is "Parallel 2-D FFT
Implementation with TMS320C4x DSPs". The code can be obtained from the TMS320
BBS ((713)274-2323).

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


Newsgroups: comp.dsp
From: spenderg@world.std.com (Stephen L Pendergast)
Subject: Wavelet loading analysis
Summary: query for wavelet transform timing
Keywords: wavelet timing loading benchmarks
Reply-To: penderga@hac2arpa.hac.COM
Organization: Hughes Aircraft Co GSG
Date: Mon, 5 Oct 1992 17:40:49 GMT

I am looking at the processing requirements for various types of wavelet
transforms including CWT of Gabor with various time and scale sampling
densities and dyadic DWT using pyramid and subband coding schemes 

I want to look at implementing them on various DSP and GP processors, so
need multiplies and adds per input sample for timing and loading estimation.
Would appreciate any references that have this type of information. Have looked
at Chui, and the various IEEE SP magazine articles, but the data would have
to be derived. Would prefer not to reimplement the wheel if its already been
done.

Thanks
Steve
 
------------------------------------------------------------------------
Stephen L Pendergast, Senior Scientist/Engineer, Hughes Aircraft Company 
Ground Systems Group, PO Box 3310 604/B152, Fullerton, CA 92634
Voice: (714)732-2579  Personal - No Company/Official Responsibility
Email: 0000440@msgate.emis.hac.com  -penderga@hac2arpa.hac.com
------------------------------------------------------------------------
From: thinman@netcom.com (Technically Sweet)
Newsgroups: comp.dsp
Subject: Re: FFT in 2D
Date: 2 Oct 92 04:25:35 GMT
Organization: International Foundation for Internal Freedom

Now, after you've taken the 2D FFT, how do you regenerate the image?

This sounds like a possible method of replicating slightly different
versions of a background texture like sand, tiny pebbles, etc.
You regenerate the waveforms but slightly perturb the FFT values 
as you go.
-- 

Lance Norskog

Data is not information is not knowledge is not wisdom.
Newsgroups: comp.dsp
From: scottc@dsr.uucp (Scott Chilcote)
Subject: Comparing C30 Development Systems
Organization: DSR
Date: Thu, 01 Oct 1992 13:47:36 GMT


Hi Folx...

My company has been using the PC Compatible C30 development system from Texas
Instruments to code the C30 in C and assembly.  We find it acceptable for most
purposes, with the possible exception of the ability to determine precise
performance time for algorithms.

We recently found out about the National Instruments development system for 
NuBUS, and are seriously considering it since we have some Mac IIcis, and the
new system costs a few thousand dollars less than the TI setup.

Is anyone familiar with the National Instruments system?  Any feedback on what
works well and what doesn't?  Our experience is that it takes some time to
determine where the pitfalls and holdups are when using one of these products,
and I'd greatly appreciate hearing from anyone who has put some mileage on one
and is willing to share the wisdom!

----------------------------------------------------------
Scott R. Chilcote
Digital System Resources
scottc!dsr.uunet.uu.net
#include <standard_disclaimers.h>

Newsgroups: comp.dsp,comp.graphics,comp.compression,sci.electronics
From: jjsmith@nyx.cs.du.edu (Jonathan J. Smith)
Subject: Re: JPEG/MPEG compression/decompression chips?
X-Disclaimer: Nyx is a public access Unix system run by the University
	of Denver for the Denver community.  The University has neither
	control over nor responsibility for the opinions of users.
Keywords: JPEG, MPEG
Organization: Nyx, Public Access Unix at U. of Denver Math/CS dept.
Date: Mon, 5 Oct 92 17:54:59 GMT

In article <1992Oct5.041450.6355@usage.csd.unsw.OZ.AU> brianm@plod.cbme.unzw.oz.au (Brian Murray) writes:
>I have been told of the existence of JPEG/MPEG compression/decompression
>chips for (in particular) real-time video with limited data bandwidth.
>Does anybody know what these chips are, who makes them, and what they
>are capable of (thats if they exist at all ...).
>
>Please respond by email.
>Many thanks,
>------------------------------------------------------ BM
>brianm@cbme.unsw.edu.au
>

There are two companies that have working and available chips for MPEG compression/decompression.
I can give you referances or email you the information if your interested.

Jonathan Smith
From: phil@tukey.berkeley.edu (Phil Lapsley)
Newsgroups: comp.dsp
Subject: CD <-> DAT sample rate conversion
Date: 6 Oct 92 03:12:22 GMT
Organization: U.C. Berkeley -- ERL
NNTP-Posting-Host: tukey.berkeley.edu

I'm posting the following for somebody who doesn't seem to have netnews
posting capabilities.  Please respond to the address below, not to me.
He'll summarize the info and I'll add the summary to the FAQ, since this
comes up periodically.  (In fact, I recall Tom Parks and somebody else
exchanging thoughts about how to build filters for this application, and
somebody mentioning a sample rate conversion filter design program
somewhere, but I don't recall where).

Phil Lapsley		phil@ohm.berkeley.edu		...!ucbvax!ohm!phil

-----

From: kaamran@sun14.vlsi.uwaterloo.ca (Kaamran Raahemifar)
Newsgroups: Comp.dsp
Subject: Sampling rate converters (from 48kHz to 44.1kHz and vice versa)
Distribution: na
Organization: University of Waterloo


Hello everybody,

(Sorry if this has been asked before)

I need to design a sampling converter for compact disk to digital audio tape
and vice versa. If you please kindly provide me with these information:

  1)-  Where can I get the coefficients of those filters? 
  2)-  Where can I get some useful information on this subject?.
       -Is there any way to get rid of that ugly filter?(Like to
        casecade some realizable ones.If so, with what specification).
       -What would be the final problems(like signal to noise ratio,...)
  3)-  Where can I get some demo or simple(or big!!) source file(s)?!.

Any help (even one piece of trash) would save me some time. Any help
would be appreciated. Thanks in advance.

kaamran@sun14.vlsi.uwaterloo.ca
Newsgroups: comp.dsp
From: feiten@sdec1.zrz.tu-berlin.de (Bernhard Feiten)
Subject: Optimizing frequency resolution with DFT
Reply-To: feiten@sdec1.zrz.tu-berlin.de (Bernhard Feiten)
Organization: ZRZ TU-Berlin
Date: Wed, 30 Sep 1992 14:47:18 GMT

--
-----------------------------------------------------------------------
|  TU-Berlin,Inst.f.Kommwiss,Sekr.H51, Dr.-Ing. Bernhard Feiten       |
|  Strasse des 17.Juni 135,D-1000 Berlin 12   Tel.: +49 30 314 25699  |
|  INTERNET: feiten@mvax.kgw.tu-berlin.de                             |
-----------------------------------------------------------------------
Hi all. 

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.

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 ? 

We think that pitch synchronous analysis with variable window size could be a 
solution. I'm searching for literature or somebody having experience on that
topic ?

Newsgroups: comp.dsp
From: mchen@groucho.cs.psu.edu (Michael Chen)
Subject: Re: realtime dsp with soundcards
Followup-To: comp.dsp
	<39074@sdcc12.ucsd.edu>
Date: Tue, 6 Oct 1992 19:48:16 GMT

In article <39074@sdcc12.ucsd.edu> shaehnic@sdcc13.ucsd.edu (Steve - Happy Hacker!) writes:

   > 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

   > 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) 

   BTW, does anyone know of a lower-end PC soundcard that CAN do
   simultaneous ADC/DAC?  Most of the common ones can't seem to.

According to the soundcard roundup that Keyboard magazine did, there's a slew
of new cards coming out... if you have a little patience, you may find what
you want.  Personally, I'm waiting for the Ensoniq sound card that has 32
voices, 1 MB of voice ROM (I think?), and dual stereo (4-track) CD-quality
playback and recording for $349.

And the Ensoniq card is supposed to be able to do simultaneous playback and 
recording at that speed. 

I feel better about Ensoniq (a REAL synth maker with a rep for insane price/
performance value and good stuff) than Gravis (a joystick maker???), so in my
book Gravis loses on this buy.
--
_____________________________________________________________________________
| Michael Chen			| From the depths of our most lucid horrors |
|				| spring our fond hopes and pure desires... |
| mchen@groucho.cs.psu.edu	| except what comes from HELL! :-) 7/23/92  |
\_______________________________\___________________________________________|
Newsgroups: comp.dsp
From: johna@comm.mot.com (John Arens)
Subject: Re: FFT in 2D
Reply-To: johna@comm.mot.com
Organization: Motorola
Date: Mon, 28 Sep 1992 21:17:14 GMT


In article <1992Sep22.163256.15935@versyss.com>, keithd@versyss.UUCP (Keith Doyle) writes...
>I'm interested in using FFT for noise reduction of images, and while I
>have worked with FFT in 1D (for audio) and have an assortment of FFT
>source code for 1D, I don't really know how to apply it to the 2D case.
>Is it simply an FFT of all the rows combined with an FFT of all the columns
>of pixels in the image?  

>>    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.
>>
>>
>>                  Hope this is useful info,
>>
>>
>>                                Dave Hall

Dave's description of the 2-D FFT is accurate.  However, the process requires 2N 1-D FFT's NOT N^2.  Taking the FFT of each row requires N 1-D FFT's.  Then, the FFT of the columns requires an additional N 1-D FFT's.  Resulting in 2N 1-D FFT's.

John Arens
johna@comm.mot.com
Newsgroups: comp.dsp
From: vjh21@cas.org (Vince Herried)
Subject: HC16 int_test
Organization: Chemical Abstracts Service
Date: Mon, 28 Sep 1992 12:46:39 GMT

Hi all.  Anyone besides me having trouble getting the interrupt
demo program to work using autovector6.?
I'm feeling confident that I've filled in the '?' marks correctly
but the interrupt code never gets called.  I put in a couple of
extra lines of code in the interrupt routine to increment a byte
of memory each time it gets called and nadda...  zip.

I've down loaded some example programs from the motorola bbs,
one of which uses the pid and also autovectors.  In their
example they do not use autovector 6.  They use the first
'user' vector at x'70'.


I got the qspi sample program to work first time.
Am struggling with the interruprt and peak programs now.  Since
I'm not getting the periodic interrupt timer to work.

Vince
-- 
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
				 |     Vincent Herried  (614) 447-3600 x 2877
				 |     Chemical Abstracts Service
 _____________________           |     P.O. Box 3012
(_)________________   \          |     Columbus, OH 43210-0012
  ________________|\   \         |
 (_)______________\_\   \        |     Current thought: People who cannot be
   ______________________\       |     understood are geniuses, so management
  (_)____________________|       |     assigns them the toughest projects.
				 |
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
From: sjreeves@eng.auburn.edu (Stan Reeves)
Newsgroups: comp.dsp
Subject: Re: Optimizing frequency resolution with DFT
Date: 2 Oct 92 15:05:56 GMT
Article-I.D.: fourier.sjreeves.921002100555
Organization: Auburn University Engineering

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.
>
>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 ? 

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.


--
Stan Reeves
Auburn University, Department of Electrical Engineering, Auburn, AL  36849
INTERNET: sjreeves@eng.auburn.edu
Newsgroups: comp.dsp
From: warner@rhoads.etdesg.trw.com (Jeff Warner)
Subject: Re: A simple question
Organization: TRW, Inc., Redondo Beach...
Distribution: usa
Date: 2 Oct 92 22:28:31

   mchung@ee.tamu.edu (Maranatha Chung) writes:

   From: mchung@ee.tamu.edu (Maranatha Chung)
   Organization: Texas A&M University
   Sorry if this has been asked before.

   Why is an odd order FIR filter better than an even order FIR filter?
   Thanks.

It is not, both are equally good.  The only difference is the group
delay.  For an odd length fir filter the group delay is (N-1)/2, note
that this results in a integer number of sample delays.  For an even
length filter the group delay is also (N-1)/2, but the delay is NOT
integral.  If this makes even length filters worst that odd then I
guess this answers your question...


Jeff Warner
Newsgroups: comp.dsp,comp.sys.ibm.pc.soundcard
From: smintz@sedona.intel.com (Stephen Mintz exp 11/92~)
Subject: Re: realtime dsp with soundcards
Originator: smintz@alamo
Organization: Intel Corporation
Date: Tue, 29 Sep 1992 15:06:48 GMT


The Altec Lancing acs-300 units have a built in DSP system in the
speakers...does, that help?
Stephen
From: croley@r2d2.mcc.com (David Croley)
Newsgroups: comp.dsp
Subject: Classic RLS filter Code
Date: 1 Oct 92 14:57:57 GMT
Article-I.D.: mcc.1992Oct1.145757.17823
Reply-To: croley@r2d2.mcc.com (David Croley)
Organization: MCC CAD Program


I am looking for C code for a classic RLS filter implementation.  If it
already exists out there, it would save me some time.  Any help would
be appreciated.

David Croley
croley@mcc.com