HDTenEightyP/NetNews huggingface dataset:
From: jay@cmn19.Stanford.EDU (Jay Kadis)
Subject: Re: RF interference
Organization: DSO, Stanford University
Date: Thu, 5 Nov 92 22:39:57 GMT

Have you experimented with a Faraday cage or some other kind of  
electromagnetic shielding?  I used to troubleshoot electrophysiology  
recording setups where microelectrodes of several hundred megohm impedance  
were the microphone-equivalent.  In order to locate sources of EM  
radiation, I made a simple little antenna out of a 1000 megohm resistor  
that was shielded with a metal housing except for one side which was left  
open.  The ends of the resistor were connected to a 100 MHz differential  
oscilloscope (even with a high Z-in you'll lose a lot of signal, but it  
doesn't matter if you have enough gain) and the display was observed as  
the antenna was moved around the room.  You might be able to do something  
like this to see if the RF is directional or not...If the RF can be  
detected in every direction, your power distribution system may be at  
fault.  Maybe the metal structure of the buildings are not grounded,  
allowing the buildings themselves to become a re-broadcast system.  Please  
let us know what you find out...


Jay Kadis
jay@ccrma.stanford.edu
Newsgroups: rec.audio.pro
From: jay@cmn19.Stanford.EDU (Jay Kadis)
Subject: Re: Whatever happened to biasless recording
Organization: DSO, Stanford University
Date: Tue, 10 Nov 92 19:20:17 GMT

In article <BxIDnI.2st@cen.ex.ac.uk> PVossler@cen.ex.ac.uk (P.J.Vossler)  
writes:
 
> Was that the system where there is a cubic relationship between the
> audio signal and the magnetic flux generated in the head gap. I recall
> reading about a system which removed the need for a high frequency
> bias signal by using such a relationship in the recording process. The
> system was supposedly capable of achieving the quality of a CD with
> simpler analog electronics.
> 
> However .....
> 
> IMHO no big company will no implement an analog tape system that
> produces CD quality sound, both Phillips & Sony have invested millions
> in developing their new systems for digital tape - I imagine any
> patents on a new analog system will have been bought up and locked
> away under heavy guard.
> 
> As an electronics buff I'd be interested in any further pointers on  
this.
> 


Yes, it is the cubic relationship between magnetic flux amplitude and the  
remanent flux that is supposed to explain this biasing scheme.  The system  
works by taking the cube root of the audio signal up to a threshold level  
which corresponds to the saturation level of the tape.  Above the  
threshold, the signal is linear.  No special playback electronics are  
required, but the playback time constant needs to be changed due to a 6  
dB/octave rise in amplitude generated by the system.

As I see it, the problems are two-fold: a cube root circuit is far from  
simple to implement accurately and with low noise and the system can only  
record on virgin tape, since it does not provide an AC bias erase head.   
This may be enough to explain it's obscurity: according to Studio Sound,  
Philips has passed on the offer from Pettigrew because it "did not reach  
our expectations".  

Jay
jay@ccrma.stanford.edu
Newsgroups: rec.audio.pro
From: bill@bilver.uucp (Bill Vermillion)
Subject: Re: Whatever happened to biasless recording
Organization: W. J. Vermillion - Winter Park, FL
Date: Wed, 11 Nov 1992 05:13:09 GMT

In article <1992Nov10.192017.14666@leland.Stanford.EDU> jay@cmn19.Stanford.EDU (Jay Kadis) writes:
>In article <BxIDnI.2st@cen.ex.ac.uk> PVossler@cen.ex.ac.uk (P.J.Vossler)  
>writes:

>> Was that the system where there is a cubic relationship between the
>> audio signal and the magnetic flux generated in the head gap.

>Yes, it is the cubic relationship between magnetic flux amplitude and the  
>remanent flux that is supposed to explain this biasing scheme.  The system  
>works by taking the cube root of the audio signal up to a threshold level  
>which corresponds to the saturation level of the tape.  Above the  
>threshold, the signal is linear.

What???  The signal can NOT be linear above the saturation level.

At saturation there is no increase in output for any further increase
in input.  How in heck can this be linear.   Once you saturate you lose
all ability to discriminate any signal change.   Or is there something
magic about this cube root recording that throws the laws of physics
right out the window.

Or - conversley - am I completely clueless?     I am willing to learn
but this doesnt' seem at all logical.

-- 
Bill Vermillion - bill@bilver.oau.org  bill.vermillion@oau.org
                - bill@bilver.uucp 
                - ..!{peora|tous|tarpit}!bilver!bill
From: gmw1@cunixa.cc.columbia.edu (Gabe M Wiener)
Newsgroups: rec.audio.pro
Subject: Do we need an FAQ?
Date: 12 Nov 92 14:54:55 GMT
Article-I.D.: news.1992Nov12.145455.28976
Organization: Columbia University

I was just wondering what the general consensus is on whether it may be
time for a rec.audio.pro FAQ.  It just seems that things like "What is
bias?" and "How do I record [X] with hardware [Y]" seem to pop up a lot.

Comments, anyone?


--
Gabe Wiener - Columbia Univ.     "This 'telephone' has too many shortcomings 
gmw1@cunixa.cc.columbia.edu       to be seriously considered as a means of 
N2GPZ in ham radio circles        communication. The device is inherently of
72355,1226 on CI$                 no value to us." -Western Union memo, 1877
Newsgroups: rec.audio.pro
From: fdeck@dumpster.helios.nd.edu (francis deck)
Subject: Miking a string bass
Organization: OUC, University of Notre Dame
Distribution: us
Date: Fri, 6 Nov 1992 20:46:47 GMT

I am looking for a microphone to use on stage with my string bass.
I'm typically playing in small jazz combo groups.  Here are some
of my problems/requirements:

1. Want balanced output, and preferably no batteries, though I'd 
   be willing to use a battery-powered mike if the battery life
   were long.

2. I want to be able to turn my amplifier way up before experiencing
   feedback.  What type of mike would give characteristics like
   this?

3. A big problem is that the bass is the quietest instrument in
   the group, and all the attempts I've made at miking it
   resulted in so much pickup of the drum set that it was
   a big failure.  The mike has to be very close to the bass,
   and that probably affects the polar characteristic of
   any mike, right?

4. Cheap and rugged!

Can anyone offer me advice?

f.deck
physics
notre dame
From: michaels@tekig7.PEN.TEK.COM (Michael L Schaeffer)
Newsgroups: rec.audio.pro
Subject: Re: MD vs. DCC: What does the group think?
Date: 9 Nov 92 17:15:53 GMT
Reply-To: michaels@tekig7.TEK.COM
Organization: Tektronix, Inc., Beaverton,  OR.

In article <1992Nov6.202431.28496@odin.corp.sgi.com> archer@elysium.esd.sgi.com (Archer Sully) writes:
>Eventually some magazine or another will do some serious testing and 
>we'll get something that more closely resembles information.
>-- archer
I have seen a couple of fairly technical articles on DCC in AUDIO magazine.
Sept 91 contains a description of the compression tehnique, while April 92
describes a listener test, and the tests performed by the sceptical author.
Both indicate that DCC will be superior to analog casettes, but will
contain some audible artifacts. Someone else here mentioned some MIX
articles on MD, which I at least will now seek out, as I would prefer MD
over DCC for the purpose of archiving my record collection in a simple to
use format.


-- 
===================================================================
michaels@tekig7.PEN.TEK.COM                          ,---c
			      #[Tek]----------------===-     Probe
===================================================================
Newsgroups: rec.audio.pro
From: zerobeat@intacc.uucp (Ferenc Szabo)
Subject: Re: MD vs. DCC: What does the group think?
Date: Tue, 3 Nov 1992 04:55:13 GMT
Organization: Inter/Access' Matrix BBS

>In article <68502@cup.portal.com> SteveSgt@cup.portal.com (Steven Eugene Sergeant) writes:
>>1.      Which format appeals to you more as a consumer?  Why?

DCC appeals more because of compatibility with regular cassettes and the
fact that EVERY review I've read of both formats says that DCC sounds
wonderfully perfect, while MD is an obvious compromise on sound quality.

>>2.      Which format would you rather master a project on?  Why?

I currently use DAT for mastering and I'm quite happy with it.  Needless
to say I'd pick DCC over MD any day.  

>>3.      Which format do you predict will be more popular?

Nobody argued that BETA was better than VHS in the early 80's.  Nobody
argues that point at anytime, but the fact that more VHS movies got
rented (and made (??)) by the mass public drowned BETA.  I suppose
more people nowadays care about sound quality than before...so I can
only hope that DCC wins out.  I certainly hope that massive public 
awareness that DCC sounds exactly like CDs but MDs are somewhat
crappier might kill the MD.

>>4.      What are your impressions if you've had a chance to demo a
>>        prototype of either format?

Not heard either; only read 6 or 7 reviews in mags I kinda trust.

Why the hell couldn't they make an MD that sounds like CDs??????
It's a more convenient format than tape, and convenience might
just overpower superior sound in a lazy person's market.

ferenc
From: chrisc@gold.gvg.tek.com (Chris Christensen)
Newsgroups: rec.audio.pro
Subject: Re: Balanced vs. unbalanced
Date: 6 Nov 92 22:08:12 GMT
Organization: Grass Valley Group, Grass Valley, CA

In article <1992Nov3.065543.18259@news.weeg.uiowa.edu> 
smills@news.weeg.uiowa.edu (MuffinHead) writes:

>   Thanks everyone for the useful and concise explanations! If the revamped
>EQ were our main EQ, I'd check the rest of the components and possibly put
>balanced jacks back in and rework all the cabling. But, it's just our
>monitor EQ and only has about 3' of wire attached to it.

It's not "balanced" connector 'cause you can do that on a three
circuit 1/4" phone plug..... I just wanted that to be clear... It's
the balanced circuit.

>To move on to something else...

>   Whenever possible, we run lights from a 220 box. When there isn't a 220
>hookup available, we have to put up with dimmer buzz through the mains.
>Besides trying to isolate circuits for lights and audio, is there anything
>else that can be done to purge the 60Hz buzz dimmer packs are so fond of?

This problem is usualy causes by dimmer noise on the grounds, If you
are using true balanced signals you can lift pin 1 on the feed from
the mixer to the amp rack at the amp rack.

I work with sound and lighting power from the same power distribution
panel and I don't have any problems any more since I lifted the pin 1
at the amp rack.

The above discussion will only be a soultion if the signals used are
truely balanced and the equipment in the amp rack is truely a balanced
input.

The use of a three pin audio connector doesn't always mean that the
circuit is truely balanced.

"truely balanced"= electronic or transformer balanced.

-- 
Chris Christensen                  The opinions I express are my own,
chrisc@gold.gvg.tek.com            and sometimes they are wrong!
916-478-3419 FAX 916-478-3887      After all, I *AM* only human.
From: jimh@ultra.com (Jim Hurley)
Newsgroups: rec.audio.pro
Subject: Info request for monitor headphones, AKG K400
Date: 11 Nov 92 21:22:09 GMT
Article-I.D.: ultra.1992Nov11.212209.13464
Organization: Ultra Network Technologies

Can anyone suggest some reasonably good monitoring headphones for, say,
under $200?. I prefer the ear pads that surround rather than touch the
ear, for those longer sessions.

I'd also prefer them to be, well, euphonious, rather than analytic, for
lack of a better term.

I do have a good reference that I use mostly, the Stax Lambda Pro, but
I wanted another pair that I can beat around and carry with me.

I am leaning towards the AKG K400, although I haven't listened to
terribly many yet. These have a mellow high end, but great detail, and
a bit of a mid-bass rise that sounds nice. I wouldn't call them
accurate, but they seem to be perfect for extended listening, without
fatiguing the ear.

They also seem to have the kind of earpad that won't decay into
dust in 4 years.
-- 
Jim Hurley --> jimh@ultra.com  ...!ames!ultra!jimh  (408) 922-0100
Ultra Network Technologies / 101 Daggett Drive / San Jose CA 95134
From: evansmp@uhura.aston.ac.uk (Mark Evans)
Subject: Re: Balanced vs. unbalanced
Organization: Aston University
Date: Thu, 5 Nov 1992 10:35:44 GMT

grego@bit.UUCP (Greg Sanguinetti) writes:
: In article <456@bit.UUCP> grego@bit.UUCP (Grego Sanguinetti) writes:
: >.... The shield is connected to the negative signal line at one end
: >and left unconnected at the other. Any interference picked up by the shield is
: >lost since it is not connected at both ends.
: 
: OOPS! ERRATA: I should have said that the shield is connected to ground at
: one end NOT the negative signal line. Sorry. Oh by the way not all cords
: have open shields since most balanced systems provide ground lift switches.

The one end should also be the signal source end.
-- 
-------------------------------------------------------------------------
Mark Evans                                   |evansmp@uhura.aston.ac.uk
+(44) 21 565 1979 (Home)                     |evansmp@cs.aston.ac.uk
+(44) 21 359 6531 x4039 (Office)             |
From: jrusby@darkwing.uoregon.edu (Jim T. Rusby)
Newsgroups: rec.audio.pro
Subject: Re: Recording interviews?
Date: 13 Nov 92 19:15:39 GMT
Article-I.D.: pith.1e0ussINNn1d
Organization: University of Oregon Network Services
NNTP-Posting-Host: darkwing.uoregon.edu

In article <1992Nov13.133830.17044@panix.com> moskowit@panix.com (Len Moskowitz) writes:
>Stephen R. Savitzky writes:
>
>>Any suggestions on how to record an interview?  One person asking
>>questions, one answering, the result to be put on cassette and
>>(eventually) sold.  What sort of microphones are good, and in what
>>arrangement?  DAT or 4-track?  How are these things usually done in
>>the broadcasting world?
>
>If your listeners will be playing the interviews back over headphones,
>you can't do any better than using a set of binaural microphones.
>Binaurals mount in or near your ears (typically clipped to the
>sidepieces of your glasses) and record exactly what your ears hear.
>When played back over headphones it sounds like you're there.  The
>sense of spatial location is phenomenal.
>
>If you combine the binaural microphone set with a DAT recorder you'll
>make very high quality recordings.
>
>Obligatory plug: I build a few different models of good sounding,
>reasonably priced, miniature in-ear and near-ear binaural microphones.
>Details will be sent on request.
>
>
>-- 
>Len Moskowitz
>moskowit@panix.com

There is only one down side to this approach, and its not a fault of the mic.
Your interview room or space will add its own character to the recordings.
If you have background noise
of any kind you'll need to deal with it.  The other suggestions of PZMs
will be affected by this too.

Now I like binaural recordings ....But....
Binaural sounds great over headphones. Speakers are another matter.
Sometimes you will get a hole in the middle. What is your likely customer
for your product? Will they listen in mono, stereo, on walkman, tv,radio etc?
 Whatever method you decide on, listen to your test recordings in that
format to determine its suitability.

I use an SASS-P for interviews - Its a sorta binaural (but not really) mic
that uses pzm elements.I do not recommend buying one. It Works fine, but I
need to be sure the room is
quiet , and the ones on each side are quiet. If our interview time takes
place when the building is busy, we use two EV re15 s to close mic the
parties and then gate it out later. NOt the best way, but Im not in
control of the interview schedule.......

A friend of mine used a set of Lens' microphones for an interview and they
really did a nice job.  The only problem he had
was an air conditioner in
the next room. Fortunately he had the time to do some noise gating before
the piece went on the air. 

jrusby@darkwing.uoregon.edu
uo school of music
Newsgroups: rec.audio.pro
From: jay@cmn19.Stanford.EDU (Jay Kadis)
Subject: Re: the usual silly test levels question
Organization: DSO, Stanford University
Date: Thu, 12 Nov 92 18:31:11 GMT

In article <1992Nov12.094326@kuttner.sfc.sony.com>  
weaver@kuttner.sfc.sony.com (Eric Weaver) writes:
> 
> Okay, here's the general answer to what "0 dB" means...
> 
> THe reference level depends on what kind of machine / audio standard  
you're
> using.
> 
> Pro systems tend to used balanced 600 ohm interconnects, and the  
reference
> level is traditionally 1 milliwatt (the "m" in "dBm"), or .775 V RMS.   
Many 
> pro systems set their VU meters to read "0 VU" at +4 dBm, except for  
some
> reason broadcasters like to set it even hotter, at +8 dBm probably to  
get
> a little edge on RF interference.
> 
> Consumer systems use a reference of 1 volt RMS ("dBv" or "dBu"), and  
tend to 
> run their meters zeroed at -10 dBv.
> 
> Now why is 600 ohms a standard?  I am fairly sure it was the  
characteristic
> impedance of old-fashioned open wire phone lines.  Most interconnect  
cable
> is actually in the 100 ohm range these days.  
> 
> Hope this helped!
> -- 


Actually, the correct reference levels are denoted:

0 dBu = .775 volt (dBv is the same, but not the preferred notation)
0 dBV = 1 volt
0 dBm = 1 milliwatt (regardless of circuit impedance: at 600 ohms, this 
	happens to be .775 V)

(See Sound Recording Handbook by John Woram)

This stuff is more confusing than US tax law...

Jay
From: jrusby@darkwing.uoregon.edu (Jim T. Rusby)
Newsgroups: rec.audio.pro
Subject: RF interference
Date: 5 Nov 92 17:49:55 GMT
Article-I.D.: pith.1dbms3INNl3g
Distribution: world
Organization: University of Oregon Network Services
NNTP-Posting-Host: darkwing.uoregon.edu


Here is something that has been bugging me.  Here at the UO campus, we
have certain buildings that seem to be famous  for on-site RF resonance in
any audio devices they may contain.

This was the first problem I tackled when I was hired as the recording eng
at the School of Music.  I now have a small library of info on rf, emf,
etc control - and of course there is no single answer to the problem.
Back when we recorded with a flown pair of 414's it became necessary to
use both isolation xfmrs on the power to the board and an rf "trap" at the
board input. Because our recording chain was of marginal quality, the
"trap" did not degrade signal quality to any noticable degree. (Please,
lets not debate this one. This recording chain had SERIOUS deficiencies
that made an rf trap a moot point.)

The Soundfield mic I am now using ignores RF beautifully, so I quit trying
to worry about a problem I no longer have. I even recorded silence and
normalized it in SoundDesigner to check for RF- its not there. I no longer
use rf traps.

Our recital hall contains a small recording booth constructed in the late
70's, no longer used for recording.  Any recording device or microphone
placed in this room picks up country western loud and clear. This is true
whether or not the tape deck, mic, and preamp are all run off ac/or
batteries.

Changing any of the conditions around a simple recording chain in this
room has no effect. I carried an Otari 5050 up there, plugged it in, put
on headphones and listened to Merle Haggard without any tape rolling!

This is the first time in 15 years I have not been able to find a simple
answer to this sort of problem. In the other places I've worked there has
always been things to change, be it star grounding, or open grounds.
Is it possible that the power serving this room is not grounded properly
and is creating a giant resonant circuit at this freq? 

This condition exists in a few of our other older buildings, its not that
uncommon.  Many audio engineers in the community have been contending with
this problem for years.  Maybe we should get licensed as a re-broadcast site?

I look forward to your thoughts on this one-
jim
From: jay@cmn19.Stanford.EDU (Jay Kadis)
Subject: Re: Whatever happened to biasless recording 
Organization: DSO, Stanford University
Date: Thu, 5 Nov 92 19:03:52 GMT

For a description of Archie Pettigrew's "Contour biasing" scheme, see  
Barry Fox' article on page 44 of Studio Sound, August 1990.  I would have  
scanned and posted it, but it's 3 column format makes that difficult...

Has anyone actually heard this?

Jay Kadis
jay@ccrma.stanford.edu
Newsgroups: rec.audio.pro
From: gmw1@cunixa.cc.columbia.edu (Gabe M Wiener)
Subject: Re: the usual silly test levels question
Reply-To: gmw1@cunixa.cc.columbia.edu (Gabe M Wiener)
Organization: Columbia University
Date: Thu, 12 Nov 1992 20:03:15 GMT

In article <1992Nov12.094326@kuttner.sfc.sony.com> weaver@kuttner.sfc.sony.com (Eric Weaver) writes:
>
>Pro systems tend to used balanced 600 ohm interconnects, and the reference
>level is traditionally 1 milliwatt (the "m" in "dBm"), or .775 V RMS.  Many 
>pro systems set their VU meters to read "0 VU" at +4 dBm, except for some
>reason broadcasters like to set it even hotter, at +8 dBm probably to get
>a little edge on RF interference.

dBm is becoming a more dated standard every day.  These days, 600-ohm balanced
lines aren't quite the norm.  We're moving away from power-matched coupling
from the days of vacuum tubes and more towards constant voltage coupling, 
with impedances being ridiculously low on one end and ridiculously high on
the other.  

As such, dBm as a unit isn't all that meaningful, and for that matter, neither
is dBv.  As everything gets more and more digital, I wouldn't be surprised 
if we're in the market for another unit change.  The day will come when console
PPM's will peak out at 0 dB referenced to the maximum value of whatever the
digital word-length is.

Bob Katz gave n AES speech at a local chapter meeting some time back that
went something like "Oh where, oh where has my dBm gone!"

>Consumer systems use a reference of 1 volt RMS ("dBv" or "dBu"), and tend to 
>run their meters zeroed at -10 dBv.

WAIT!  1 dBv/1 dBu = 0.775V.  1 dBV = 1V.  American consumer gear tends to
be rated to 10 dBv as far as I've heard, though the whole concept gets
sort of corrupted in consumer gear anyway, because you never know how much
headroom you have.

>Now why is 600 ohms a standard?  I am fairly sure it was the characteristic
>impedance of old-fashioned open wire phone lines.  Most interconnect cable
>is actually in the 100 ohm range these days.  

600 ohm (dBz for those who care about such things) was the old Bell Telephone
standard, as was dBm and dBv.  600 ohms at 775 mV yields 1 mW.  Thus, when we
put the old power equation, P=V^2/R into logarithmic terms, we get dBm=dBv-dBz,
a convenient, though no longer very useful equation.

If your electronics are good enough, you really needn't worry about distorting
your amplifiers.  The big fear is slamming into 0 dB on the digital side.


--
Gabe Wiener - Columbia Univ.     "This 'telephone' has too many shortcomings 
gmw1@cunixa.cc.columbia.edu       to be seriously considered as a means of 
N2GPZ in ham radio circles        communication. The device is inherently of
72355,1226 on CI$                 no value to us." -Western Union memo, 1877
Newsgroups: rec.audio.pro
From: CZGS <CZGS@MUSICA.MCGILL.CA>
Subject: Denon DAT modifications
Organization: McGill University
Date: Sun, 8 Nov 1992 05:13:01 GMT

     Does anyone out there have any info on converting a Denon
DTR-2000 DAT recorder so that it will record with a 44.1 kHz
sampling rate from an analog input? Since the first time I asked
this group, a number of people have expresses an interest in obtaining
the same information. Please EMAIL me. Thanks.
Geoff Martin, McGill University
CZGS@musica.mcgill.ca