HDTenEightyP/NetNews huggingface dataset:
From: deh1@ns1.cc.lehigh.edu (Darrel Herbst)
Newsgroups: comp.dsp
Subject: DSP Graduate study
Date: 16 Dec 92 21:38:31 GMT
Organization: Lehigh University


    This may not be a technical post, but I'm looking for some feedback about
    graduate level university's with strong DSP programs for
    computer/electrical engineers.  I limiting my scope to the Eastern/mid
    Western United States, so if you'd like to email me about a program that
    you think is interesting, or are involved in, please feel free to do so.

    Darrel Herbst






    
-- 
           -------------------------------------------------
            Darrel Herbst : Lehigh University
            DEH1@ns1.cc.lehigh.edu
            215/758.1909
           -------------------------------------------------
Newsgroups: comp.dsp
From: ja51359@uxa.cso.uiuc.edu (axelrod)
Subject: Re: windows for FFT
Organization: University of Illinois at Urbana
Date: Sat, 19 Dec 1992 19:38:49 GMT

nyarko@ee.ualberta.ca (David Nyarko) writes:

>Hi,
>In some texts I find windows defined with an "N-1"
>denominator and in others an " N " and would like fellow netters
>to clarify this apparent discrepancy. For example if a waveform
>whose period is (N * Ts) when synchronously sampled, has it's
>frequency slightly reduced so that a period would now
>have a duration of [(N + d) * Ts], which of the following
>would be appropriate , the hanning  in a) or that in b).
>Ts = sampling period
>N is a  positive integer
>d=(0,1)

>eg
>a) In article ;"On the use of windows for the
>Harmonic Analysis with the discrete Fourier Transform"
>by Harris,

>the hanning window is defined as
>w(n) = 0.5*[1 - cos(2*n*pi/N)]  n=0,1,2,...N-1

>b) On pg. 94 of "Introduction to Applied Statistical signal
>analysis " by Richard Shiavi the hanning window is defined as:

>dt(n) = 0.5*[1 - cos(2*pi*n/(N-1))]   0 <= n <= N-1

>A few other references have either a) or b). This pattern is
>repeated for other windows too.
>When does one use a) or b).

>			David

2*n*pi/N will give you values of the window evaluated from 0 to 2pi-2pi/N,
and 2*n*pi/(N-1) will give you values of the window evaluated from 0 to 2pi.
These are assuming you are using n from 0 to N-1, thereby having N samples.
I don't know why you'd want to include the last 2pi value, since it's the 
same as the zero value and would create problems in certain cases.  I just
finished my DSP exam and am glad!!!!!!!!!!!!!!!!!!!!!!!!!
:)

Newsgroups: comp.dsp
From: mepeters@rodan.acs.syr.edu (Mark E. Peterson)
Subject: Vocoders and FFTs
Organization: Syracuse University, Syracuse, NY
Date: Mon, 14 Dec 92 23:57:46 EST

Hello. Reading through the posts on pitch shiftings I have heard nothing
about using FFTs to reconstruct voice signals using different sets of
frequencies. Is this sort of method used for vocoders? Would a 256 point
FFT be enough to do the job if you just modulated 256 certain sine waves
together according to amplitude? I would like to experiment with the
sort of idea, but I'd like to learn if it's been done before. I want to
do it in real time. I have the integer FFT posted a few days ago that
can do a real-time FFT in 2.6 milliseconds.. That combined with averaging
256 sine waves could be accomplished barely in real time at 4Khz at least.
Has anyone had or heard of any success with this?


Newsgroups: comp.dsp
From: abbott@vuse.vanderbilt.edu (Ben Abbott)
Subject: F2C port to TMS320C40           
Keywords: TI, C40, FORTRAN, F2C
Organization: Vanderbilt University School of Engineering, Nashville, TN, USA
Date: Wed, 16 Dec 1992 15:38:05 GMT


A few weeks ago I released my port of the AT&T F2C (Fortran to C) translator
for Sonitech's Spirit-40 board and the TI C compiler.  
My port includes a full I/O system via the PC bus so you do not have to 
keep creating new versions of host programs.

The release is on the TI-BBS ("f2c_soni.exe").

Has anyone tried this port ?
Would you be interested in a version of the I/O system for "C" programs ?

Ben Abbott
abbott@vuse.vanderbilt.edu
Newsgroups: comp.dsp
From: jchen@tomahawk.me.gu.edu.au (Jinghong CHEN)
Subject: Re: Matlab Comms toolbox: Postscript Doco?
Organization: Griffith University, Brisbane, Australia
Date: Wed, 16 Dec 92 08:06:15 GMT

In article <johnr.724490817@ee.uts.EDU.AU> johnr@ee.uts.edu.au (John Reekie) writes:
>Hi
>
>Has anyone managed to produce a postscript version
>of the documentation of the Matlab Communications
>Toolbox which was announced here recently? We
>can't figure out how to convert the PCL graphics.
>
>BTW, thanks to Mehmet Zeytinoglu (sp?) for releasing
>this!
>
>Regards
>
>John Reekie
>

I have the same problem.

J.CHEN
-- 
ehunjfnlkdf dfdfjnlkdfl
From: @FtCollins.NCR.COM
Newsgroups: comp.dsp
Subject: entropy
Date: 14 Dec 92 21:02:41 GMT
Reply-To: @FtCollins.NCR.COM
Distribution: usa
Organization: NCR MED, FT. Collins, CO. 80525  (303) 223-5100



Has anyone measured the entropy of real video?
(NTSC or otherwise)

Are there any published results on this?

Thanks,

Jim Prater - NCR Microelectronics, Advanced Development
jim.prater@FtCollins.ncr.com
Xref: sparky comp.dsp:2811 comp.speech:367
From: pasanen@cs.tut.fi (Pasanen Jari)
Newsgroups: comp.dsp,comp.speech
Subject: Help needed: implementation of adapt. algorithms
Date: 16 Dec 1992 13:59:13 GMT
Organization: Tampere University of Technology
Distribution: world
NNTP-Posting-Host: naakka.cs.tut.fi
Originator: pasanen@naakka.cs.tut.fi


We are about to start a research project where we implement, for instance, 
some adaptive algorithms (at least LMS) using fixed point arithmetic. 
We have done some studies on the adaptive algorithms but what seems
to be a problem now is that we don't have any good references on
the implementation issues, e.g., the effect of using fixed point in 
this context. If someone knows a good reference where these issues
are considered, we would appreciate the help. Any pointers are wellcome. 

Thank You in advance

	-pasanen-


-- 
Jari Pasanen                    # Tampere University of Technology,
pasanen@cs.tut.fi               # Research Inst. for Information Tech.
                                # PO Box 553, SF-33101 Tampere, Finland
                                # tel: +358 31 161 888  fax: 161 857
From: herman@nosc.mil (John W. Herman)
Subject:  Frequency warping software
Keywords:  FFT warping
Organization: Naval Ocean Systems Center, San Diego, CA
Date: Thu, 17 Dec 1992 01:35:44 GMT


--
Does anyone have any FFT based code to do frequency warping and
band shifting?  Thank you.
Xref: sparky comp.dsp:2832 comp.lang.c:18627
From: mrn@eplunix.UUCP (Mark R. Nilsen)
Newsgroups: comp.dsp,comp.lang.c
Subject: C to 68705 chip
Keywords: 68705
Date: 17 Dec 92 23:40:15 GMT
Organization: Eaton-Peabody Lab, Boston, MA

Hi,

I have heard of a C compiler for the Motorola 68705 family of chips.
The name I was given was ByteCraft.  I checked a few vendors but none
had heard of the product.  Is there anyone out there who has heard of
this or another C compiler for this chip?  Has a 69705 machine description
been written for gcc?  Any information would be apreciated. 

email prefered, but I will check these groups as well.

I will post a summary.

mrn%eplunix.UUCP@eddie.mit.edu

-or-

mrnilsen@athena.mit.edu

Thanks in advance.

--Mark.
-- 
Mark Nilsen.  == mrn%eplunix.UUCP@eddie.mit.edu
The world has changed and we can no longer think that logic is going to
prevail ... 
   - A. Scott Crossfield, Proceeding of X-15 first flight anniversary. 
Newsgroups: comp.dsp
From: klank@fox.nstn.ns.ca (Karen Lank)
Subject: commercial AM receivers
Keywords: AM receivers
Organization: NSTN Network Operations Centre, Nova Scotia, Canada
Date: Wed, 16 Dec 1992 20:18:33 GMT

Hello,

I am looking for some information on commercial AM receivers to aid in a
research project currently underway at Applied Microelectronics Institute.

Any information on where I might obtain a description of commercial AM 
demodulation algorithms or, ideally,  some emulation software would be greatly 
appreciated.  Automatic gain control modelling would be of particular
interest. 

Thanks,
Karen Lank.
From: bartjan@blade.stack.urc.tue.nl (Bartjan Wattel)
Newsgroups: comp.dsp
Subject: MU-law and A-law tables in the DSP56000
Date: 15 Dec 92 15:38:00 GMT
Organization: MCGV Stack, Eindhoven University of Technology, the Netherlands.
X-Newsreader: TIN [version 1.1 PL6]

Hi,

I'm starting on studying the DSP56000 for an exam at the Eindhoven University
of Technology (the Netherlands). I ran into the terms 'MU-law' and 'A-law'.
The X data ROM in the DSP56000 seems to be factory programmed with these
tables. 

Could anybody tell me in what way these tables can be used ? I have the idea
that the tables have something to do with (data)compression, but I might be
wrong...

Bartjan Wattel at Eindhoven University of Technology, the Netherlands
(Email: bartjan@blade.stack.urc.tue.nl)
--------------------------------------------------------------------------------

     "Why is it that we rejoice at a birth and grieve at a funeral?  It is
                    because we are not the person involved"
                                      -- Mark Twain
Newsgroups: comp.dsp
From: rcogs@catt.citri.edu.au (Greg Smart)
Subject: InvestiGATOR
Keywords: DSP,TMS32C30
Reply-To: rcogs@catt.citri.edu.au
Organization: CATT Centre at CITRI
Date: Thu, 17 Dec 92 02:03:51 GMT

Hi,
	I'm interested in finding out more about the InvestiGATOR
project which was publicised in Oct. 1992 issue of Signal Processing.
Is there anyone who knows the email address or the fax number of the
appropriate person at ther University of Florida.

Thanks

---
Greg Smart                              Internet: rcogs@citri.edu.au
Department of Communication and Electronic Engineering, RMIT.
Collaborative Information Technology Research Institute (CITRI)
Melbourne, Victoria, Australia.

Newsgroups: comp.dsp
From: ja51359@uxa.cso.uiuc.edu (axelrod)
Subject: Active Noise Cancellation Algorithm
Organization: University of Illinois at Urbana
Date: Fri, 11 Dec 1992 17:34:38 GMT


Hi.  I know this is possible because I've heard the military does it for
helicopter headsets; and I heard that the car companies will be doing it
too.  Does someone know of how the basics work behind active noise can-
cellation?

What I'm interested in is a system where you have a speaker or two
somehow produce a signal that is directly out of phase with the ambient
noise.  This seems impossible, since a speaker radiates in all directions,
thereby having a different phase at different points in the room.  Is 
there some way of getting an average phase or something, in order to
decrease the noise in the room/car?

I'd like references to articles on this subject, along with a simple
explanation if possible.

Thanks


Newsgroups: comp.dsp,comp.dcom.modems
From: cristja1@netnews.jhuapl.edu (John A. Cristion)
Subject: Re: Active Noise Cancellation Algorithm
Organization: JHU/Applied Physics Laboratory
Date: Tue, 15 Dec 1992 18:35:52 GMT

m22257@mitre.org (Jeff Correia) writes:

>In article <Bz3vHr.LKK@news.cso.uiuc.edu>, ja51359@uxa.cso.uiuc.edu
>(axelrod) writes:
>> 
>> 
>> Hi.  I know this is possible because I've heard the military does it for
>> helicopter headsets; and I heard that the car companies will be doing it
>> too.  Does someone know of how the basics work behind active noise can-
>> cellation?
>> 
>> What I'm interested in is a system where you have a speaker or two
>> somehow produce a signal that is directly out of phase with the ambient
>> noise.  This seems impossible, since a speaker radiates in all directions,
>> thereby having a different phase at different points in the room.  Is 
>> there some way of getting an average phase or something, in order to
>> decrease the noise in the room/car?
>> 
>> I'd like references to articles on this subject, along with a simple
>> explanation if possible.
>> 
>> Thanks
>> 
>> 

>     The type of cancellation you are referring to is one that I have heard
>most commonly called reference cancelling.  The idea is pretty much as you
>have described it.  You have one primary channel, and 1 or more reference
>channels.  The primary channel contains your desired signal corrupted by
>unwanted noise, the reference channels contain noise which is (hopefully)
>highly correlated with the unwanted noise in your primary channel, but
>contain no trace of your desired signal.  

>     A classic paper on this topic is by Bernie Widrow.  While trying to
>examine the heartbeat of a fetus, the heartbeat of the mother is a strong
>interfering signal.  Using a primary channel of mother+fetus and being able
>to locate a position to get a "clean" but maybe slightly out of phase
>refernce of the mother's heartbeat only, they were able to obtain the
>desired results.  I'm sorry I don't have the exact reference.  




There is a book "Adaptive Signal Processing," Widrow and Stearns, which deals
with this subject.  There are also many articles in the Signal Processing
and Biomedicl Engineering literature.  As Jeff has already noted the main
requirement for using the algorithms in the book is a set of independantly
measured, correlated noise signal, which may be cancelled from the reference
signal. The algorithms  also assume that the signal are linearly combined.  
If Nonlinear signal processing is needed then Neural Network models may be
needed.  Currently, here at JHU/APL, we are using Widrow's LMS algrothms 
for deriving fetal EKG signals from non-invasive measurements from the
mother.  If there is any interest, please post here or e-mail me.


-- 

_______________________________________________________________

    	John A. Cristion
Xref: sparky comp.dsp:2791 comp.dcom.modems:18034
From: lars@spectrum.CMC.COM (Lars Poulsen)
Newsgroups: comp.dsp,comp.dcom.modems
Subject: Re: DSP code for modem functions?
Date: 12 Dec 92 18:39:11 GMT
Organization: CMC Network Systems (Rockwell DCD), Santa Barbara, CA, USA

In article <2933103874.1.p00004@psilink.com>
   "Michael Smith" <p00004@psilink.com> writes:
>I'm not planning to go into the modem business, so modem companies need 
>fear no competition. The requirement is an internal one in my company -- 
>we need to put together an integrated piece of hardware incorporating 
>both modem functions and proprietary protocol and network-management 
>processing. If it's cheaper for us to license code than to write it 
>ourselves, we shall. 

I noticed that you don't say which company you work for.

I also noticed that the device you are describing sounds a lot like "a
smart modem". If the modem function is really the least part of your
system, I suggest that you simply install a Rockwell-chipset-based modem
in a corner of your circuit board.

-- 
/ Lars Poulsen, SMTS Software Engineer	Internet E-mail: lars@CMC.COM
  CMC Network Products / Rockwell Int'l	Telephone: +1-805-968-4262	
  Santa Barbara, CA 93117-3083		TeleFAX:   +1-805-968-8256
Xref: sparky comp.dsp:2839 sci.math.num-analysis:3651
From: fang@phy.duke.edu (Fang Zhong)
Newsgroups: comp.dsp,sci.math.num-analysis
Subject: Is there a MatLab user mailing list?
Date: 20 Dec 92 15:31:56 GMT
Followup-To: poster
Distribution: na


	The subject tells it all.



-- 
	Fang Zhong				1-919-660-2524
	Duke University Dept. of Physics	fang@phy.duke.edu
	Durham, N.C.      27706			
Newsgroups: comp.dsp,comp.lang.c,sci.math
From: nyarko@ee.ualberta.ca (David Nyarko)
Subject: White noise generation program
Organization: University Of Alberta, Edmonton Canada
Date: Sun, 20 Dec 1992 03:47:46 GMT

Hi,
Could I know if there is any C language program
which can generate a white-noise sequence with a
known variance. The purpose would be to test the
effect of various digital signal processing 
algorithms on signals corrupted with the noise.

		David

email responses are welcome.
From: stefank@elixir.lne.kth.se (Stefan Karlsson)
Newsgroups: comp.dsp
Subject: Adaptive filter algorithms
Date: 20 Dec 92 15:25:50 GMT
Organization: School of EE, Royal Institute of Technology, Sweden

Hello. 

Investigating adaptive noise canceling, I have trouble implementing 
some adaptive filter algorithms. 

If anyone has some program code listings in C, Pascal, Matlab, SPOX or other 
languages, please mail them to me. Information on where to find such listings
are also appreciated.

The algorithms of most interest are gradient lattice, RLS lattice and
square root RLS.

Merry Christmas

Stefan
From: johnr@ee.uts.edu.au (John Reekie)
Newsgroups: comp.dsp
Subject: Matlab Comms toolbox: Postscript Doco?
Date: 16 Dec 92 07:26:57 GMT
Organization: University of Technology, Sydney
NNTP-Posting-Host: rossi.ee.uts.edu.au

Hi

Has anyone managed to produce a postscript version
of the documentation of the Matlab Communications
Toolbox which was announced here recently? We
can't figure out how to convert the PCL graphics.

BTW, thanks to Mehmet Zeytinoglu (sp?) for releasing
this!

Regards

John Reekie
From: tom@mills.edu (Tom Erbe)
Newsgroups: comp.dsp
Subject: DSP56001 and CS4215
Date: 16 Dec 1992 07:31:32 GMT
Organization: Mills College, Oakland CA
Distribution: world
NNTP-Posting-Host: ella.mills.edu


Has anyone connected these two chips?  As I am under a deadline, I would be
forever indebted to anyone who could share their experiences with me.  DSP
or psuedo code would be very appreciated.

thank you, thank you

-- 
tom erbe 
ccm - mills college
5000 macarthur blvd.
oakland, ca 94613
Newsgroups: comp.dsp
From: kwokyuen@iti.gov.sg (Song Kwok Yuen (CCL))
Subject: G.728 LD-CELP
Keywords: LD-CELP
Organization: National Computer Board, Singapore
Date: Fri, 18 Dec 1992 09:56:16 GMT

I am looking for the latest (draft) copy of G.728, a CCITT audio
coding standard.  I would appreciate if anyone out there is able 
to send me a copy, or point me to where I could download/ftp the
standard.

Thanks and regards..

my address:
kwokyuen@iti.gov.sg

Newsgroups: comp.dsp
From: dorsey@lila.com (Bill Dorsey)
Subject: Speech Coding and PELP
Date: Tue, 15 Dec 1992 23:14:14 GMT

I have an application that requires me to encode human speech at relatively
low bit rates (<=9600bps) and not require a dedicated DSP processor.  I'm
looking for a method that could work on your typical 33MHZ 486-based PC.  I
realize that this may be a tall order, and certainly do not require that the
speech be telephone quality -- merely intelligible would suffice for starters.

I've looked into PCM, CVSD, and ADPCM, and none of these methods would seem
to work at the bit rates I'm looking for.  I briefly looked at CELP coding,
but that would seem to require more processor bandwidth than I wish to use.
Finally, PELP (Phase Excited Linear Prediction) has been mentioned as a
coding method that may work given my constraints.  A search of the local
technical bookstore has not revealed any texts that discuss this method.  If
anyone can provide details on how it works, comments on its suitability,
references to texts on it, or other suggestions for coding methods that may
meet my requirements, I will be most grateful.

-- 
Bill Dorsey      "Most people mistake law for justice and authority for
dorsey@lila.com   liberty.  You will hear them talk of 'liberty under law,'
ripem public key  and they are content to see it so deep under the law that
on request        it is completely obliterated." -- Herbert Spencer
Via: uk.ac.leeds.scs; Fri, 11 Dec 1992 17:57:30 +0000
Distribution: world
X-News-Gateway: NewsGateway@phaze.UUCP (NewsGateway 1.0 by Pierre Rioux)
Date: Fri, 11 Dec 92 17:58:12 GMT
Newsgroups: comp.dsp
Organization: The University of Leeds, School of Computer Studies
From: ross-c@scs.leeds.ac.uk
To: usenet@phaze.cam.org
Cc: 
Subject: Re: pitch shifting

In article <4055@randvax.rand.org> you write:
>In article <1351@se.alcbel.be> cgra@btma74.nohost.nodomain () writes:
>>
>>Aren't pitch shifts by large amounts _always_ going to sound strange? 
>
>This is an interesting problem, and in at least some cases there are
>workable solutions.  For example, the human voice can be usefully

[Extremely technical description deleted]

>that the formants shift as well.  However, by imposing the original
>spectral envelope on the shifted frequencies the formants can be
>maintained even though the pitch has changed.

My answer would have been "Yes, but there are things you can do to
make them sound less strange".

Cheers,

Ross-c
Newsgroups: comp.dsp
From: wang@pollux.it.lut.fi (Liuye Wang)
Subject: CFP: ICSP'93
Date: Fri, 18 Dec 1992 13:02:59 GMT
Organization: Lappeenranta University of Technology, Finland

PLEASE FORWARD THIS MESSAGE TO ANYONE WHO....
--------------------------------------------------------------
This is a call for paper of the Beijing International 
Conference on Signal Processing(ICSP'93).  The deadline 
for submission to this conference has been delayed to be 
the January 31, 1993. 

                      CALL FOR PAPER
----------------------------------------------------------------------------
IEEE International Conference on Signal Processing (ICSP'93/Beijing)
----------------------------------------------------------------------------

The second International Conference on Signal Processing will be held in
Beijing, China on October 26-30, 1993. It will include sessions on all
aspects of theory, design and applications of signal processing.
Prospective authors are invited to propose papers in any of the following
areas:

. Digital Signal Processing(DSP)
. Spectrum Estimation and Modelling
. Speech Coding
. Multidimensional signal processing
. VLSI for signal processing
. Adaptive filtering
. Array signal processing
. Underwater acoustic signal processing
. Morphological Signal processing
. Higher order spectral analysis
. Speech synthesis and recognition
. Image processing
. Pattern recognition
. AI and neural networks
. Audio and Electroacoustics

Papers dealing with topics beyond these categorized, especially with new
applications, will also be considered.

DEADLINES
  * Original deadline of Submission: November 1, 1992*

        Postphoned deadline of submission : January 31, 1993
	Notification of acceptance 	      Feburary 1, 1993
	Submission of photo-ready summary     April 1, 1993

Technical Program Committes
	H.T.Kung             U.S.A	T.S.Huang      U.S.A
	M.R.Schroeder        Germany    K.R.Rao        U.S.A
        A.G. Costantinides   UK         P.M. Grant     U.K
        Ernst Luder          Germany    K.M. Wong      Canada
	Y.Neuro              Finland    Hiroya Fulsaki Japan
        R.M. Haralick        U.S.A      Claudo Moraga  Germany
	Claude GUEGUEN       France     Yuvl V Gulyaev   Russia
    	M.Kaveh              U.S.A      W. Besslich      Germany
	Wan-Chi SIU          HongKong   Y.T.Chen         Canada
	Petros  Maragos      U.S.A      Kyu Tae Park     Korea
	M.H. Er              Sigapore   K.F. Lee         U.S.A
	T.S Durrani          U.K        D. Etter         U.S.A
      	M. Huni              U.K        S. Furui         Japan
        M.N.S. Swamy         Canada

Prospective authors are invited to submit four copies of each paper. The
abstracts should be composed of title of the paper, authors, names,
addresses and a text of about 300 words in length, and must be sent before
January 31, 1993 to:

Professor Yuan Baozong
Institute of Information Science
North Jiaotong University
Beijing 100044, China
Tel: +86 1 32.40616
     +86 1 8316622 ext. 205
Fax: +86 1 8315891

--