HDTenEightyP/NetNews huggingface dataset:
From: laut@alien.gici.com (Bill Laut)
Newsgroups: comp.dsp
Subject: Re: CD<->DAT sampling converter
Date: 11 Oct 92 21:50:52 GMT
Distribution: na
Organization: Gull Island Consultants, Inc.


     <Warning:  Slightly smart-alecy response follows...>

In article <BvxCuF.JDH@sun3.vlsi.uwaterloo.ca>, kaamran@sun18.vlsi (Kaamran Raahemifar) writes:
> 
> Hello friends,
> 
> I have got some answers from comp.dsp, and I am thankful.
> But still some questions remain:
> 
>        1: I have got some combinations for 44.1<->48 sampling converter
>           like:
> 
>   [23 lines delete]
> 
> Thank you all.
> 
> Kaamy
> 
> kaamran@sun14.vlsi.uwaterloo.ca


On the other hand, you could always do what I did.  Namely, purchase a
Sony DTC-1000ES and clip one of the diodes from the front panel.  It then
records at 44.1 just fine....  Except for stuff with the copy-prohibit bit
set, but that's another story..

It's an elegant solution:  Not much purchasing, and even less engineering...

Seriously, if there is any interest, e-mail to my address and I'll dismantle
the DTC-1000ES to remember which diode it is.  If you look for it yourself,
you can't miss it.  It's near where the Analog/Digital selector switch is,
and it's the only one there which ISN'T wave-soldered.  I mean, you can tell
this thing was put on by hand, it's such a baaaaad soldering job....

And in defference to the old Sony-vs.-CBS debacle over DAT home recording,
if what you're after is pirating CDs, then DAT is not a very cost-effective
solution.  You are much better off buying a Sony PCM-601ESD processor (which
runs at 44.1 anyway) running the CD into it, and then hooking up your favorite
VCR in extended record mode.  Six hours of CD-quality audio for the cost of
an $8.00 blank tape?  Now that's cost-effective!  And it even comes with its
own built-in "music video" :-).



-- 
Bill Laut			Internet:  laut@alien.gici.com
Gull Island Consultants, Inc.	Phone:     (616) 780-3321
P.O. Box 627
Muskegon, MI  49440		>> "Usual disclaimers, apply within" <<
Xref: sparky comp.dsp:2324 comp.sys.transputer:1112
From: paulus@cs.anu.edu.au (Paul Mackerras)
Newsgroups: comp.dsp
Subject: Re: TI 34010 PD Assembler and debugger
Date: 13 Oct 92 00:07:13 GMT
Organization: Computer Science Department, ANU, Australia

antsu@polaris.utu.fi (Antti-Pekka Virtanen) writes:

>>   Does there exists such a thing as a Public Domain assembler and/or debugger
>>   for the Texas Instrumnet 34010 chip. TI is asking $1500.00 for their

>If you really mean the TMS340 family, then all I can say is that I haven't
>located PD tools for it.

I have written a simple assembler & debugger for the TMS34010 graphics
processor, which is available as free software.  You can get it by
anonymous FTP from dcssoft.anu.edu.au, file pub/gsp.tar.Z.
The assembler doesn't have macros and doesn't produce relocatable
code, and (unfortunately) there isn't any documentation to speak of yet.
However, it comes as source, and if anyone would like to extend it
or write documentation, and contribute that, that would be very welcome.

Paul Mackerras
paulus@cs.anu.edu.au
Newsgroups: comp.dsp
From: rayw@ee.ubc.ca (raymond w m woo)
Subject: wavelet analysis -- where to start?
Organization: University of BC, Electrical Engineering
Date: Sun, 11 Oct 1992 07:36:53 GMT

Hi,

  I would like to know more about the theory of wavelet analysis,
especially Fast Wavelet Transform (I have no knowledge in this area
at all right now).  Can someone pls tell me which tech. papers I should read?
Thanks in advance!  

-- 
::::::: Raymond WM Woo  <rayw@ee.ubc.ca> ::::::: Elec. Eng. UBC, Canada :::::::
From: pj_pycro@csd.uwe.ac.uk (PJ Pycroft)
Newsgroups: comp.dsp
Subject: FFT/DFT progs wanted...
Date: 9 Oct 92 15:45:44 GMT
Organization: University of the West of England, Bristol

Hello netters,

I'm in the progress of setting up a project on noise recognition systems,  
recognize some form of noise such as siren (police, fire etc.), alarm
signal or any other warning noise which will alert a deaf person.

I'm will be using Signal Processing and Neural Network for detecting
above noise and will be using parallel processing such as Transputers.
I won't be using DSP chip's cos' of the limitation of our facility
but Transputers will do the job....

I would be grateful if you could help me ... where can I find the FFT
as well as DFT programs (128, 256, 512 points and perhaps more!!.)
in 'C' source code and if there is any possibility for me FTP' these
programs.

And, any other info' on above subject would be greatly appreciate...

I'm a profoundly deaf postgraduate student doing M.Sc. in Parallel Computing
Systems.

Thanks for your help.

Paul

---------------------------------------------------------------------------
Internet: pj_pycro@csd.uwe.ac.uk         | The Transputer Centre,
          pj_pycro@csd.uwe-bristol.ac.uk | University of the West of England,
                                         | Bristol, BS16 1QY. U.K.
---------------------------------------------------------------------------
Newsgroups: comp.dsp
From: gcouger@olesun.okstate.edu (Gordon Couger)
Newsgroups: comp.dsp,sci.electronics
Subject: Re: Cheap 12-bit A/D board for PC
Date: 15 Oct 92 03:35:50 GMT
Article-I.D.: osuunx.1992Oct15.033550.4893
Distribution: na
Organization: Oklahoma State University, Stillwater

In article <1992Oct14.230003.12839@unislc.uucp> mgc@unislc.uucp (Marc Christensen) writes:
>Kevin Gans (gans@ee.ualberta.ca) wrote:
>: Hi,
>: Could anyone indicate sources for a cheap 12-bit A/D or
>: data acquisition board for an IBM or compatible 386 PC.
>: A minimum of 2 channels would be required and a minimum
>: sampling rate of 44ksamples/sec.
>: Email responses are welcome.
>: 				Kevin
>Try some of the Newer sound cards they are getting cheaper and some can do 
>2 input ie stereo at 44khz using 16 bit resolution.
>
>Marc Christensen
> 
 I have used several a/d converters and a Soundblaster card on several different projects and the sound blaster is by far the best.

Gordon Couger
From: rpool@next9.cs.umr.edu (Ryan Pool)
Newsgroups: comp.dsp
Subject: A4000 Sound (was Re: Motorola's DSP 56001 and 96002)
Date: 15 Oct 92 02:52:26 GMT
Article-I.D.: umr.1992Oct15.025226.9888
Organization: University of Missouri-Rolla, Missouri's Technological University

> >The laatest amiga , the A4000, does _NOT_ have a built in DSP... 
> >>--
> 
> Altough Commodore plans to release a AT&T DSP32 board next spring.   
Let's
> hope it will also fit in current Amigas (at least in the 3000 :)

Commodore has also discussed their future plans in detail for the first  
time ever.  Upgrading the sound to match the upgraded graphics is their  
first priority now.  I have heard rumors on the nets that the DSP support  
in the new system will blow everything else out of the water.  But then  
that's just a rumor and we don't want to spread those around now do we?

>    __ Skipping every ?86 numbered lines in your code will remove all  
your
> __/// remaining bugs and bottlenecks 8^)
> \XX/  Benoit W Sevigny, software engineering student,  
sevigny@info.polymtl.ca


I like this ^^^^^^^^^^^^^^^^^^^^^
From: etl0@ns1.cc.lehigh.edu (ERIC TODD LANDRIEU)
Newsgroups: comp.dsp
Subject: Help with TMS320x20 AIB
Date: 14 Oct 92 18:27:29 GMT
Organization: Lehigh University

After looking through the manuals, I have found nothing mentioning any way of
triggering an interrupt.  Here's the situation:
   1) I'm creating a system that will change the frequencies of an audio
   signal musically.  Thus, I might bring everything up an octave, or a
   semitone, or whatever.  I have the theory for that all down.
   2) I want to add hardware logic(using gates and F/F's) that will allow the
   user to change the number of semitones shifted.  The user presses the up or
   down button, which would send a trigger to the interrupt.  Meanwhile, the
   flip-flops would be used to hold the current amount the pitch is shifted,
   while displaying it on (either one or two) 7-segment LEDs.  This part is no
   problem.
So, I just basically want to know if any means are possible to trigger an
interrupt in the system like this, and what addresses would need to be changed
for setting the address of the interrupt routine, the masks, etc.
                        Thanks for any help.
                        Eric Landrieu

                        
-- 

               ===============================================
               =   Eric Landrieu  =  ETL0@ns1.CC.lehigh.EDU  =
               ===============================================
"I'd be apathetic, but I don't really care about it.
From: jpcampb@afterlife.ncsc.mil (Joe Campbell)
Newsgroups: comp.dsp
Subject: Re: wavelet analysis -- where to start?
Date: 12 Oct 92 22:47:23 GMT
Organization: The Great Beyond

In article <1992Oct11.073653.483@ee.ubc.ca> rayw@ee.ubc.ca (raymond w m woo) writes:
>Hi,
>
>  I would like to know more about the theory of wavelet analysis,
>especially Fast Wavelet Transform (I have no knowledge in this area
>at all right now).  Can someone pls tell me which tech. papers I should read?
>Thanks in advance!  
>
>-- 
>::::::: Raymond WM Woo  <rayw@ee.ubc.ca> ::::::: Elec. Eng. UBC, Canada :::::::

I highly recommend W. Press' recent unpublished, but ftp'able, work:
%0 Report
%A Press, W.
%D 1991
%T Wavelet Transforms (to appear in Numerical Recipes: The Art of Scientific Com
puting, 2nd ed.)
%I Harvard-Smithsonian Center for Astrophysics
%8 July 21
%9 Preprint
%@ 3184
%X Pedagogical review
%O From: Bill Press, Harvard University.
Date: Tue, 21 Jul 92 11:40:31 -0400
Subject: Numerical Recipes software
 
This fall, the second edition of the Numerical Recipes book will come
out. It will include a chapter on wavelets and software both in Fortran
and C.
 
A preliminary version of the paper and the software can be downloaded
from anonymous ftp to 128.103.40.79, in /pub/wavelet.tex and
/pub/wavelet.f.
The figures are, unfortunately, NOT available on-line, and there are
no more preprints.

Joe
-- 
.............................................................................
; Joe Campbell  N3JBC  jpcampb@afterlife.ncsc.mil  74040.305@compuserve.com ;
; My opinions are mine!   Happiness = Reality - Expectations, Click & Clack ;
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Newsgroups: comp.dsp
From: rolfe@fraser.sfu.ca (Christopher John Rolfe)
Subject: Re: Vocal Erasers - What can we leave in?
Organization: Simon Fraser University, Burnaby, B.C., Canada
Date: Sat, 17 Oct 1992 00:03:45 GMT

ross-c@scs.leeds.ac.uk (R Clement) writes:

>Since bass is usually at lower frequencies than vocals, would it not
>be possible to filter out the bass parts of the stereo signal (hopefully 
>avoiding nearly all of the vocal) and then add this back to the (monoised)
>vocal cancelled signal. This would (theoretically) add some stereo width
>back to the signal.

>Can anyone suggest a suitable frequency cutoff (in Hertz) for a low pass
>filter to filter out the bass parts of a music signal?

>What about a high pass filter, what are the highest frequencies present in
>a lead vocal? Are they low enough so that there's useful stuff in the higher
>frequency ranges (I'm using 48Khz sampling, and processing output from a
>CD player).

	The low E-string on an electric bass is usually tuned to about
41 Hz., a bit less than an octave below a bass singer.  Male pop
singers seem to be predominantly tenors or baritones which means that their low
note fundamental would be up about an octave from that.  So you may
find two octaves more or less between the bass and the vocal. 
	Problem is that the overtones of the bass are probably quite
necessary to its overall sound and distinctiveness, up to 160 Hz. and beyond.
There's probably also a lot of other program information in the mid-range.

	What you might try is putting 
the mono version of the original unprocessed tune onto a
track A and B of a two track tape.  Then dub the double mono version
thusly:
	A ---> voice-eraser -_
                             _ ---> mono result
	B ---> filter ------

	Use a band-pass filter on B, set the lower cutoff freq. to 0,
and adjust the upper cutoff by ear for the optimal result ( perhaps
between 80 and 160 Hz.).  
	I don't expect this to be very transparent, but if you track
the song a bit ( with the upper cutoff ) you might get something
useable. 


-- 
Chris                   | 2617 East 4th,| If brevity is the soul of 
rolfe@sfu.ca            | Van., B.C.    | wit, what is semibrevity
Simon Fraser University | V5M 1K5       | the soul of?
Burnaby,  Canada        | Canada        | 
Newsgroups: comp.dsp
From: leung@sce.carleton.ca (Tze-Wo Leung)
Subject: Vector Linear Prediction
Keywords: LPC,Levinson-Durbin
Organization: Carleton University
Date: Thu, 15 Oct 1992 15:53:13 GMT

Hi, everyone:

I would like to find out if anyone has any experience with the
design of vector linear predictor.  I understand there is a
generalized version of the Levinson-Durbin algorithm for
finding the coefficient matrices for vector linear predictors
in the general case of m-th order statonary k dimensional vector
sequences.  I would much appreciate if someone who might have 
written a C routine for doing this can share the code with me.
My interest is in predictive coding of LPC parameter vectors
for speech signal.



Thanks,
Tze-Wo Leung
-------------------------------------------------
Chinese Name (in .HZ encoded format): ~{A:WS:M~}
Addr.: Dept. of Systems & Computer Engineering,
       Carleton Univesity, 
       Ottawa, Ontario, Canada K1S 5B6.
Voice: 613-788-2600 Ext. 1948  Fax: 613-788-5727
Email: leung@sce.carleton.ca
-------------------------------------------------
--
Best Regards,
Tze-Wo Leung
-------------------------------------------------
Chinese Name (in .HZ encoded format): ~{A:WS:M~}
From: warwick@cs.uq.oz.au (Warwick Allison)
Newsgroups: comp.dsp
Subject: Re: Motorola's DSP 56001 and 96002
Date: 12 Oct 92 05:16:15 GMT
Reply-To: warwick@cs.uq.oz.au

loj@cae.wisc.edu (Lo Jeffrey) writes:

>Sorry for asking a dumb question, but what is a falcon030 and 040?

The falcon030 is the new low-cost machine from Atari Corp. - it has a
16MHz 68030 and a 32MHz 56001 (with 32K of 24bit words), excellent (16bit)
sound, and graphics (VGA + 65000 colour True-Colour), all BUILT-IN.

Probably a cheaper platform for DSP development than any card for IBM's or
Apple Macs - the falcon030 is priced at $800 for the bare machine (no
harddrive, no monitor, only 1 megabyte - add any SCSI drive, or 2.5" IDE drive,
and any VGA/SVGA monitor).

It looks like DSP is finally coming to home computers (NeXT already have
them, but they're STILL out of my price range).  I hear the latest Amiga
also has DSP.  Now we might start seeing home applications - who'd bother
in the past when machines did not come standard with DSP.

--
Warwick
--
  _-_|\      warwick@cs.uq.oz.au            /Disclaimer:
 /     * <-- Computer Science Department,  /  
 \_.-._/     University of Queensland,    /   C references are NULL && void*
      v      Brisbane, Australia.        /  
Newsgroups: comp.dsp
From: parnoux@dorsai.com (Patrick Arnoux)
Subject: TI 34010 PD Assembler and debugger
Organization: The Dorsai Embassy +1.718.729.5018
X-Newsreader: Tin 1.1 PL4
Date: Fri, 9 Oct 1992 02:44:21 GMT


Does there exists such a thing as a Public Domain assembler and/or debugger
for the Texas Instrumnet 34010 chip. TI is asking $1500.00 for their
development set, and that is definitly not cost effective for a one off
project. Any and all info or leads are appreciated.
	Thanks in advance

			Patrick Arnoux
			(parnoux@dorsai.com)

Newsgroups: comp.dsp
From: loj@cae.wisc.edu (Lo Jeffrey)
Subject: Re: Motorola's DSP 56001 and 96002
Organization: U of Wisconsin-Madison College of Engineering
Date: 9 Oct 92 11:14:19 CDT

In article <Bvt76B.CMt@ufrima.imag.fr> mathevet@boole (MATHEVET Jerome) writes:
>I'm willing to purchase the falcon030 as soon as it is released .
>The 030 version is to have the DSP 56001 and the 040 the 96002 (the 
>latest DSP from Motorola). Unfortunately, noone has ever edited anything
>about them (In france). If anyone has the motorola's programmer handbook
>in a file (the complete specs, if available), it would be nice to mail
>them to me (I mean, E-mail of course).
>

Sorry for asking a dumb question, but what is a falcon030 and 040?

Jeff Lo
loj@cae.wisc.edu
Newsgroups: comp.dsp
From: g9024557@cc.uow.edu.au (James Phillip ANDREW)
Subject: Re: Optimizing frequency resolution with DFT
Organization: University Of Wollongong
Date: Sun, 11 Oct 92 04:10:29 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.

dak@messua.informatik.rwth-aachen.de (David Kastrup) writes:
>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. 
[..]

The probability density function described above must be positive and sum 
(integrate) to 1: is therefore the normalised square of the (time domain) 
function an appropriate measure (in a mean square error sense) ? 

i.e. 
variance = SUMi { [xi-mean(xi)]^2 * f(xi)^2 } 
where SUMi ( f(xi)^2 ) = 1.

Also , does anybody  know of some good references on this subject ?

James Andrew
Wollongong
g9024557@wampyr.uow.oz.au
From: fa@dcs.ed.ac.uk (Fuzz Ahmad)
Newsgroups: comp.dsp
Subject: help with DSPs
Date: 13 Oct 92 16:34:43 GMT
Organization: Department of Computer Science, University of Edinburgh

Hello, I am new to this newsgroup and I was wondering if anyone out
there could help me.

I am building a DSP board based around a Motorola DSP96000 to process
sound signals.  I will be using it to implement a surround system and I
need some information on how to create effects such as converting your
living room to sound like a cathedral, jazz club, stadium, etc. and
other such effects - something similar to these 'digital sound field
processors' you can buy today.

I also need to get hold of a C compiler for the 96000.  Does anyone know
where I can pick one up for free?

Thanks very much in advance,

Fuzz.

TF
Newsgroups: comp.dsp
From: kaamran@sun18.vlsi (Kaamran Raahemifar)
Subject: CD<->DAT sampling converter
Originator: kaamran@sun18.vlsi.uwaterloo.ca
Organization: University of Waterloo
Distribution: na
Date: Sat, 10 Oct 1992 21:25:26 GMT


Hello friends,

I have got some answers from comp.dsp, and I am thankful.
But still some questions remain:

       1: I have got some combinations for 44.1<->48 sampling converter
          like:

             3:2,  7:5, 7:16 or
             7:8,  3:5, 7:4  or
             3:10, 7:4, 7:4 

          The true factorization gives us:

           44.1   147     3*7*7
          ------=------=-----------
            48    160   2*2*2*2*2*5


          If your answer is yes, which one is better?.
              
                       In the book of
            ( Multirate Digital Signal Processing     
               By  Crochiere and L.R. Rabiner),
               They do not mentioned the way of optimizing the process
               for L:M in general. They only discuss for Decimator or 
               Interpolator.(Not both together)

       2: What kind of filters would be in most efficiency in this 
               application?.
          Windowed ones, polyphase ones, IIR ones , ...?

Thank you all.

Kaamy

kaamran@sun14.vlsi.uwaterloo.ca
Xref: sparky comp.dsp:2380 sci.electronics:17251
From: tadams@wedge.sbc.com (Tom. Adams 529-7860)
Newsgroups: comp.dsp,sci.electronics
Subject: Re: Cheap 12-bit A/D board for PC
Date: 16 Oct 92 13:29:14 GMT
Distribution: na
Organization: Southwestern Bell Technology Resources, St.Louis, MO

>In article <1992Oct14.230003.12839@unislc.uucp> mgc@unislc.uucp (Marc Christensen) writes:
>>Try some of the Newer sound cards they are getting cheaper and some can do 
>>2 input ie stereo at 44khz using 16 bit resolution.

In article <1992Oct15.033550.4893@osuunx.ucc.okstate.edu> gcouger@olesun.okstate.edu (Gordon Couger) writes:
> I have used several a/d converters and a Soundblaster card on several different projects and the sound blaster is by far the best.

Is there software to use Soundblasters as general A/D convertors?  How is
this typically set up?  What are the limits on input voltages and such.
--
Tom Adams	tadams@sbctri.sbc.com adams@swbatl.sbc.com   314-529-7860
misology (mi-sohl'-ogee) Hatred of reason, argument, or enlightenment.
--  misologist n.	Not in any way connected to the word management.
From: ross-c@scs.leeds.ac.uk (R Clement)
Newsgroups: comp.dsp
Subject: Extracting right/left signals from a stereo signal
Date: 12 Oct 92 23:06:38 GMT
Organization: The University of Leeds, School of Computer Studies
Originator: ross-c@csparc2

Hi.

I'm trying to work out how to implement both stereo enhancers and 
vocal erasers (in a possibly mistaken view that they are related).

I'm trying to extract "pure" left and right signals from a stereo signal.
Assuming the siginal I've got has something panned hard right, something
panned hard right, and something in the centre, I can write the sum of
the left and right channels as:

	raw(left)+raw(right) = true(left) + true(centre) + true(right)

when summed to give a mono signal.

Now, if I calculate: raw(left) - raw(right)

I get: 		     true(left) - true(right)

Where the centre signal is absent, and the left and right signals are
present relatively out of phase (this may be useful as a vocal eraser of
poor quality).

But I have been unable to use combinations of sums and differences of 
raw(left) and raw(right) to isolate the pure left and right signals. To
describe what I mean by "pure" I mean that something panned exactly 
centre would be absent, but something panned 10% to the right would be 
present only in the "pure right" signal (i.e. not at all in the pure
left signal) and would be attenuated proportional to its centrality
(i.e. dead centre = 0% original volume, halfway between centre and right
= (e.g.) 50% original volume, hard right 100% original volume).

Ummmm, I hope I've made myself clear here. Is there any simple way of
doing this?

Thanks in anticipation,

Ross-c
Newsgroups: comp.dsp
From: sassoj@lib106.its.rpi.edu (John J. Sasso Jr.)
Subject: NEED REFS FOR DSP ALGs
Organization: Rensselaer Polytechnic Institute, Troy NY
Distribution: usa
Date: Thu, 8 Oct 1992 14:23:05 GMT

Hi!

   Can anyone refer me to books and/or journals which contain source code
(preferably C) for DSP filtering techniques, or just DSP in general?
I am trying to find code for adaptive filters, and I have looked in Numerical
Recipes in C, but came up empty.

                   John


Newsgroups: comp.dsp