HDTenEightyP/NetNews huggingface dataset:
Newsgroups: bit.listserv.emusic-l
Date:         Mon, 5 Oct 1992 12:37:23 +0100
From:         Martin Roth <roth@IPS.ID.ETHZ.CH>
Comments: cc: yourmessageofsun@rothin-reply-to.BITNET, 55+0000.@04oct9221.BITNET

>Ok the anaolgy kind of breaks down there, but if you think of a pure sound
>say a sine wave (or a high flute if you like) This would normally be a
>continuous curve. Digitizing makes it stepped. The more bits, the smaller
>the steps. Well in one direction (amplitude). sampling rate steps in the
>time domain. So...

>What one hears is actually a kind of distortion in reality. In one bit
>sampling things would either be on or off, so a sine wave would become
>square. most distortion is about curves becoming square. In the worst
>cases 8-bit sounds like a cassette recorder being driven too hard!

>let's face it - 8-bit is crap

Well, the question about 8 vs. 16 bit interested me, but so far I have not
seen any good aswers!
The sine-wave is a good exaple, by digitizing it you get no longer one sine
wave but a step-like wave with the rough apperance of a sine. Sure 16 bits
steps are a better approximation to the original, but WHY? Sampling Therem
states that disortions caused by the sampling in discrete time steps causes
only disortions higher in frequency than half the sampling frequence. So if
you add a low-pass after the DAC (what is done allways), you have NO distortion.

If I think about it, I believe that Sampling Theorem is true for discrete-step
sampled continuous signals, but assumes continous values at each step.
Is there a sort of extension to reduce the single sample steps to a finite
number of levels? How does that affect frequency response?

To illustrate my questions, imagine a 2-level digitised sound at an extremely
high sampling rate. I'm sure it can not reproduce anything like music, but
why? How can the Sampling Theorem be extended to take into account the
discrete signal levels?

Could _this_ be the diffrerence between a good vinyl record and a
cd (I know some people who state that vinyl, on a high end
player of course, is still a lot better than top-quality cd).

-Martin
_______________________________________________________________________
      _   Martin Roth                Martin Roth    ETHZ, ips, RZ F16
|\  /|_)  Mail: roth@ips.id.ethz.ch  Sandacker 14   g 01/256 55 68
| \/ | \                          CH-8154 Oberglatt p 01/850 32 75
          (Eng. Comp. Sci. ETH)      Switzerland    (F-)emails welcome!
-----------------------------------------------------------------------
From: xrjdm@CALVIN.GSFC.NASA.GOV ((Joseph D. McMahon))
Newsgroups: bit.listserv.emusic-l
Subject: Version 1.1a of LIME (notation software) now available -
              comp.music #4604
Date: 29 Sep 92 16:59:20 GMT
Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Comments: To: emusic-l@auvm.american.edu

Reply-To: xrjdm@calvin.gsfc.nasa.gov
In article <BvAKEF.FIA@news.cso.uiuc.edu>, cottle@cmp-rt.music.uiuc.edu (David
 M. Cottle) writes:
>
> Hi,
>
> A demonstration version of Lime 1.1a is available for Macintosh
> SE and II series computers.  A binhex'ed stuffit file containing
> the executable and manual can be copied via anonymous FTP from
> /pub/lime on novamail.cerl.uiuc.edu (internet number:
> 128.174.180.9). Read the "README" file in /pub/lime for further
> instructions.
>
> The demonstration version is fully functional, but is limited to
> three pages.
>
> Here are the most important differences between 0.7 and 1.1:
>
> 	-Several pieces can be edited at  one time.
>
> 	-Several pages  from the same piece can be edited at one
> 	 time.  The pages can be from the same or different notation contexts.
>
> 	-You can copy and paste between several pieces you are
> 	 editing at one time.
>
> 	-You can enlarge or reduce (zoom) the view.  (See Layout.)
>
> 	-It supports Sonata and Interlude fonts.  (See Options.
> 	 You must purchase these fonts separate from Lime.  )
>
> 	-System breaks can be set at fixed (always the same) or
> 	 variable (calculated by the program) locations.
>
> 	-Unusual accidentals (user defined, also some quarter
> 	 tone symbols that can be adjusted in cents on synthesizers
> 	 that play microtones) are available under the note menu.
>
> 	-Dashed ties and slurs
>
> 	-"X moves with note" and "Y moves with note" have been
> 	 changed to "Horizontal lock" and "Vertical lock."  NB:
> 	 Having the boxes checked now has the opposite effect.
> 	 Horizontal Lock checked means the annotation does not
> 	 move left or right with the note, but is "locked" in
> 	 position on the page.  Likewise, having the Vertical
> 	 Lock checked means the annotation will not move up and
> 	 down with the note, but is locked on the page.
>
> 	-The Parameters dialogue has been improved, making
> 	 it easier to find the parameters.
>
>
>
> If you have problems or questions, please write Email to the Lime
> Consultant.  If you decide you would like to buy Lime after you
> experiment with it, the ordering information is in the front cover
> of the manual, or call (217) 244-4298.
>
>
> Lippold Haken
> CERL Sound Group
> 252 ERL -- University of Illinois
> Urbana, Il   61801-2977
>
> Lime Consultant:   Lime@uiuc.edu
>
> (dmc)

--
Joe McMahon |     "Those crusty old stodgers wouldn't know an exciting idea
P.O. Box 13 |        if it picked up a lowpass filter and four-poled them."
Lothian, MD 20711 --------------------------------------------------------------
xrjdm@twinpeaks.gsfc.nasa.gov
Newsgroups: bit.listserv.emusic-l,bit.listserv.allmusic,alt.emusic,rec.music.newage
From: Mike J. Brown <mjbrown@magnus.acs.ohio-state.edu>
Subject: Ambient FAQ available
Organization: The Ohio State University
Date: Mon, 5 Oct 1992 06:51:37 GMT
X-Mailer: ELM [version 2.3 PL11]

The first draft of the Ambient FAQ is finished.  I posted it to
rec.music.info, alt.rave, and rec.music.industrial.  Readers of
other newsgroups/mailing lists can check the places I posted it
or I can email you a copy (65K).

Mike
Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Newsgroups: bit.listserv.emusic-l
Date:         Mon, 5 Oct 1992 12:39:41 +0100
From:         Martin Roth <roth@IPS.ID.ETHZ.CH>
Subject:      Re: 8/16 bit

>Ok the anaolgy kind of breaks down there, but if you think of a pure sound
>say a sine wave (or a high flute if you like) This would normally be a
>continuous curve. Digitizing makes it stepped. The more bits, the smaller
>the steps. Well in one direction (amplitude). sampling rate steps in the
>time domain. So...

>What one hears is actually a kind of distortion in reality. In one bit
>sampling things would either be on or off, so a sine wave would become
>square. most distortion is about curves becoming square. In the worst
>cases 8-bit sounds like a cassette recorder being driven too hard!

>let's face it - 8-bit is crap

Well, the question about 8 vs. 16 bit interested me, but so far I have not
seen any good aswers!
The sine-wave is a good exaple, by digitizing it you get no longer one sine
wave but a step-like wave with the rough apperance of a sine. Sure 16 bits
steps are a better approximation to the original, but WHY? Sampling Therem
states that disortions caused by the sampling in discrete time steps causes
only disortions higher in frequency than half the sampling frequence. So if
you add a low-pass after the DAC (what is done allways), you have NO distortion.

If I think about it, I believe that Sampling Theorem is true for discrete-step
sampled continuous signals, but assumes continous values at each step.
Is there a sort of extension to reduce the single sample steps to a finite
number of levels? How does that affect frequency response?

To illustrate my questions, imagine a 2-level digitised sound at an extremely
high sampling rate. I'm sure it can not reproduce anything like music, but
why? How can the Sampling Theorem be extended to take into account the
discrete signal levels?

Could _this_ be the diffrerence between a good vinyl record and a
cd (I know some people who state that vinyl, on a high end
player of course, is still a lot better than top-quality cd).

- -Martin
_______________________________________________________________________
      _   Martin Roth                Martin Roth    ETHZ, ips, RZ F16
|\  /|_)  Mail: roth@ips.id.ethz.ch  Sandacker 14   g 01/256 55 68
| \/ | \                          CH-8154 Oberglatt p 01/850 32 75
          (Eng. Comp. Sci. ETH)      Switzerland    (F-)emails welcome!
- -----------------------------------------------------------------------


------- End of Forwarded Message
Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Mail-System-Version: <VAX-MM(284)+TOPSLIB(151)@UNHH.UNH.EDU>
Newsgroups: bit.listserv.emusic-l
Date:         Sat, 3 Oct 1992 10:25:37 -0400
From:         Dana Nibby <D_NIBBY@UNHH.UNH.EDU>
Subject:      MA, NH
Comments: To: emusic-l@auvm.american.edu, rec-music-synth@ucbvax.berkeley.edu,
              bpm+@andrew.cmu.edu

  Does anyone besides myself read these lists in the University of New
Hampshire area? (Durham, Dover, Somersworth, Portsmouth, etc) If so,
I'd like to correspond with you. Drop me a line with what type of music
you do and what gear you have.


   Pax,

    Dana
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Subject: (no subject given)
Content-Length: 948
Content-Type: text
Original-From: wbf@alux2 (Fox)
Original-To: att!auvm.american.edu!emusic-l
Newsgroups: bit.listserv.emusic-l
Date:         Mon, 5 Oct 1992 11:12:00 EDT
From:         wbf@ALUX2.ATT.COM
Comments: To: emusic-l@auvm.american.edu.att.com

Subject: 8 vs. 16 bits

For those who are interested, the dynamic range of a digital system may
be expressed by:

                                             n
                        DR in dB = 20 log ( 2  - 1 )

where n is the number of bits.  Therefore, a 16 bit system has a
theoretical limit to its dynamic range of approximately 96.3 dB and an 8
bit system's limit will be about 48.1 dB.  As you can see, a violin
playing a dynamic of mf or p will have some overtones being more than 48
dB below the digital system's maximum output volume.  In the 16 bit
system, these overtones are captured where the 8 bit system looses it
altogether.  Even a "bad" analog system would reproduce the overtones
better than the 8 bit digital system.  The sound may be in the noise
floor, but, like listening to an old 78, the ear has an amazing ability
to hear through the noise at least a little bit.  (No pun intended, but
I'll take what I can get!)

Bill
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Return-Path: crispen <@ada3.ca.boeing.com:crispen@efftoo>
X-Envelope-To: EMUSIC-L@AUVM.AMERICAN.EDU
Newsgroups: bit.listserv.emusic-l
Date:         Mon, 28 Sep 1992 08:05:47 CDT
From:         crispen <crispen@EFFTOO.BOEING.COM>
Subject:      Re: Think C-MIDI i/o
Comments: To: EMUSIC-L@ada3.ca.boeing.com

>>what the heck is a semaphone?
>
>A mis-spelled semaphore.

And a semaphore is, of course, two.
+-------------------------------+--------------------------------------+
| Rev. Bob "Bob" Crispen        |            "IT FOUND ME!"            |
| crispen@foxy.boeing.com       |      Campus Crusade for Cthulhu      |
+-------------------------------+--------------------------------------+
Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Subject: (no subject given)
Via: uk.ac.birmingham.computer-centre.vax1; Sun, 4 Oct 1992 22:43:23 +0000
Newsgroups: bit.listserv.emusic-l
Date:         Sun, 4 Oct 1992 22:50:00 LCL
From:         DOWRJ@VAX1.COMPUTER-CENTRE.BIRMINGHAM.AC.UK
Comments: To: EMUSIC-L%AUVM.AMERICAN.EDU@BIRMINGHAM.AC.UK

It is probably easier to think in terms of graphics when it comes to the
problems of 16-bit v's 8-bit sound. When there are only 256 values being
represented (8-bit) everything is a bit grainy, the more bits, the less
things seem 'digitized'. 16-bits gives you 65536 different values and so
changes are less abrupt. If one thinks of a picture represented by 1 bit
i.e. there is only black *or* white (no grey scale) there is really very
little detail there, one only gets an outline. The more steps there are
the more subtly one sees, more detail. In sound one 'hears' as it where
more detail.

Ok the anaolgy kind of breaks down there, but if you think of a pure sound
say a sine wave (or a high flute if you like) This would normally be a
continuous curve. Digitizing makes it stepped. The more bits, the smaller
the steps. Well in one direction (amplitude). sampling rate steps in the
time domain. So...

What one hears is actually a kind of distortion in reality. In one bit
sampling things would either be on or off, so a sine wave would become
square. most distortion is about curves becoming square. In the worst
cases 8-bit sounds like a cassette recorder being driven too hard!

let's face it - 8-bit is crap
Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Subject: (no subject given)
X-Mailer: Mail User's Shell (7.2.3 5/22/91)
Newsgroups: bit.listserv.emusic-l
Date:         Mon, 5 Oct 1992 13:33:48 +0100
From:         "(Adam MIROWSKI)" <mir@OPUS.CHORUS.FR>
Comments: To: Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>

> I believe that Sampling Theorem is true for discrete-step
> sampled continuous signals, but assumes continous values at each step.
> Is there a sort of extension to reduce the single sample steps to a finite
> number of levels? How does that affect frequency response?

These are linear systems, so the initial-quantization error propagates
simply as an additional signal. I don't think it affects the frequency
response.

> To illustrate my questions, imagine a 2-level digitised sound at an extremely
> high sampling rate.

Easy. It is right in front of me.

> I'm sure it can not reproduce anything like music, but why?

So this George Duke CD is not music. OK. Your opinion. :-)

> How can the Sampling Theorem be extended to take into account the
> discrete signal levels?

By lowering the cutoff frequency of the reconstruction filter.

> Could _this_ be the diffrerence between a good vinyl record and a
> cd (I know some people who state that vinyl, on a high end
> player of course, is still a lot better than top-quality cd).

When people have spend a fortune on those HE players, no
wonder they will never tell it wasn't worth it... :-)
From: mico@milano.oas.olivetti.com (Michele Costabile)
Newsgroups: bit.listserv.emusic-l
Subject: Re: 8/16 bit
Date: 5 Oct 92 21:45:55 GMT
Reply-To: mico@milano.oas.olivetti.com (Michele Costabile)
Organization: Olivetti Research California

In article <CH40MAILQUEUE-101.921005141214.832@oak.scp.ac.uk>,
M.Rootes@SHEFFIELD-CITY-POLY.AC.UK (Martin Rootes) writes:

> I often doubt what Hi Fi buffs say about pieces of equipment etc, there are
> many reasons why records sound different from CDs. Theorectically a perfect
> record player will be more accurate than a perfect 16bit CD, as record
players
> are continuos analog and so do not suffer from quantisation and digitisation
> noise. However, there is no such thing as a perfect record cutter, let alone
> a perfect record player, perfect amp and perfect speakers. CD's probably give
> a more accurate output than even top of the range record players, but some
> people prefer the softer (less high frequencies) sound of records.
Oh, no, better not start this thread.  Turntables can and do sound as clear
as you like and as accurate as you like: listen to one of the best and
you will hear the difference.
If the concept of a good turntable is some direct drive Japanese thing
then I will agree with you, if you want it to cost less than $300 (yes,
the turntable) then CDs do sound better.
On the high end turntables sound really good, everybody can hear it.
The CD anyway does sound really MUCH better in the price range where
turntables are hissy, scratchy and bad sounding.
An other good thing is that since every owner of a cheap CD can tell
that it sound better, the companies sell the same music at twice the
price with the same costs i.e. they make more money.
Polygram is owned by Philips and CBS by Sony, this is why you do not
find records any more, they don't want to make everything double or triple.

Michele

       "if you have a cache miss, you can make a check ... or charge"
From: ricard@axis.se (Ricard Wolf)
Newsgroups: bit.listserv.emusic-l
Subject: Re: Think C-MIDI i/o
Date: 28 Sep 92 08:39:05 GMT
Organization: Axis Communications AB, Lund Sweden

In article <9209211134.aa29922@dcs.ed.ac.uk> Nick Rothwell <nick@DCS.ED.AC.UK> writes:
>>what the heck is a semaphone?
>
>A mis-spelled semaphore.

Hm. Hey, this could be the name of a new instrument! Some sort of
two-sound synthesiser? Two-positioned synthesiser? Ideas please! ( ( :-) )

/ "Ricard Wolf on Semaphone"
-- 
Ricard Wolf                   / | \  / | /-           email: ricard@axis.se
Axis Communications AB       /__|  \/  | \__          uucp:  axisab.se!ricard
S - 223 70 LUND             /   |  /\  |    \         Tel:   +46 46 19 18 63
SWEDEN                     /    | /  \ | \__/         Fax:   +46 46 13 61 30
Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Newsgroups: bit.listserv.emusic-l
Date:         Mon, 28 Sep 1992 10:48:22 EDT
From:         Jeff Jacobucci <JACOBUJJ@DUVM.BITNET>
Subject:      Re: Electronic Music School

please unsubscribe me from this magazine.
Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Mailer: Elm [revision: 66.25]
Newsgroups: bit.listserv.emusic-l
Date:         Sun, 4 Oct 1992 21:58:09 EDT
From:         Keith Hall <khall@S850.MWC.EDU>
Subject:      Re: 8bit vs 16bit?
Comments: To: "(Electronic Music Discussion List)" <EMUSIC-L@AMERICAN.EDU>

>
>  Can anyone tell me what the difference is in sound quality between 8bit
> and 16bit? I understand about sampling rate, and that if the rate is too low,
> aliasing/clipping occurs. I've asked this question before and gotten no
 respon.

First, quality of sound (good, bad) is usually a very personal thing.
Sure a synth puts out a noisey hissy sound it could be considered bad,
but not everybody hates that (especially if you are interested in the
technique involved in creating Techno music).

As William Flanagan explains the technical difference is the density of
the digitized material.  As you know, a sample which has a sampleing
rate of say 40k (for numerical conveinience) is simply sampling a signal
40 thousand times a second.  This actual sampleing is convertind the
amplitude of the wave, which in electerical terms is simply the amount
of electricity moveing through the cable hooked into the input of the
sampler.  Each sample is a number which represents a relative amplitude.
If you only have 256 possible values, then man different voltages will
be represented as the same value, meaning that the definition of a wave
will be limited to the larger amplitude difference and not the smaller.
All this means that lower amplitude signals (high frequency is usually
at a lower amplitude) will not have as much definition in the wave.
This usually means that you will never get the truly bright sound of a
crash symbol of a high hat.  But hey, I have settled for 12-bit for a
year now and with a 32k sampling rate.  But its a Roland S-550 which is
a truly incredible machine(an earlier version of the 770 which should be
on everybodies wish list).  If you are really interested in Techno
musich I would advise working with a fullbodied sampler an not use the
computer, it will make your life a thousand times easier and will amaze
you with the sound quality difference.


Keith Hall
khall@s850.mwc.edu
Newsgroups: bit.listserv.emusic-l
From:  jvm@medicine.msk.su (Savitsky Vladimir Vladimirovich)
Subject: Question about sound boards
Date: Tue, 29 Sep 1992 08:21:17 GMT
Reply-To: jvm@medicine.msk.su
Organization: Independent Research Company J.V. Medicine Inc.

     One small  Russian company is highly interested in
any information  about dealers  which are  selling  the
computer sound  boards and  MIDI interface  boards  for
IBM-compatible  hardware like MT-32, Pro  Audio, AdLib,
SoundBlaster etc. If anybody can send  us addresses and
names of these companies, we would be very thankfully.
     Also we  would rather  prefer the German companies
but any information will be appreciated.  Please  write
us to address jvm@medicine.msk.su.

     It's our  first letter  into  this  conference  so
sorry for possible misunderstanding


     With great respect       V. Savitsky

Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Newsgroups: bit.listserv.emusic-l
Date:         Mon, 5 Oct 1992 13:37:49 EDT
From:         The Radio Gnome <V2002A@TEMPLEVM.BITNET>
Subject:      Re: Techno....

Hi,

     One problem with Techo/E-Music in general is the proliferation of
easy technology.  With built in sounds and plug 'n play modules, just
about any idiot can make 'music'.  The result has been a lot of stuff with
the same patches, the same samples, etc...  Even with state of the art
machinery it still takes time to learn the stuff.  All to often, nobody has
the patience.  I first realized this when my brother bought a Korg
DW-8000.  A lot of the stuff I was hearing on alternative radio was filled
with sounds that the DW had been shipped with, little or no variation.

     I myself have forgone big $ samplers and keyboards proper.  The music
I make is done entirely with cheap digital delay samplers, analog tape,
the effects rack from my PA and a variety of bizarre treatments.  I get all
my samples off air or from cable TV.
     With the demise of the Experimental Radio Project here in Philadelphia,
I've lost my only outlet along with numerous other local artists.

                         Andy Wing
Comments: Gated by NETNEWS@AUVM.AMERICAN.EDU
Newsgroups: bit.listserv.emusic-l
Date:         Mon, 5 Oct 1992 13:16:40 EDT
From:         The Radio Gnome <V2002A@TEMPLEVM.BITNET>
Subject:      Re: 8/16 bit
              5 Oct 1992 12:39:41 +0100 from <roth@IPS.ID.ETHZ.CH>

For a good continuation of this thread, check out the past discussions on
AUDIO-L@VMTECMEX.  We've already churned over LP vs CD and 16/18/24/32 bit
sound and data compression and other various other topics of interest.

     A general thought is that no matter how good the digital media may look
in theory, the circuitry itself remains essentially analog.  Conversion is
always a weak link.

                          Andrew Wing
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Newsgroups: bit.listserv.emusic-l
Date:         Tue, 6 Oct 1992 20:06:00 +0000
From:         Nick Rothwell <cassiel@CASSIEL.DEMON.CO.UK>
Subject:      8-bit Ratners
Comments: To: Electronic Music Discussion List <EMUSIC-L@american.edu>

>It is probably easier to think in terms of graphics when it comes to the
>problems of 16-bit v's 8-bit sound. When there are only 256 values being
>represented (8-bit) everything is a bit grainy, the more bits, the less
>things seem 'digitized'

This is not necessarily a bad thing. 8-bit Grunge is good in places. Listen
to the PPG.

>let's face it - 8-bit is crap

Well, it works for me.

Nick.
From: mico@oliveb.ATC.Olivetti.Com (Michele Costabile)
Newsgroups: bit.listserv.emusic-l
Subject: Re: Think C-MIDI i/o
Date: 28 Sep 92 21:45:58 GMT
Organization: Olivetti ATC; Cupertino CA, USA

What's a semaphone?
Easy: the semaphone reads ASCII text and with the help of a series
of conversion tables translates semantics of the text it read into
music.
Better ideas sound better, weird ideas sound experimental, traditional
ideas play a major triad.
Sometimes no sound is produced.


Michele

       "... and after shut down a shut up may sometimes be necessary"
Newsgroups: bit.listserv.emusic-l
From: mclemens@silver.ucs.indiana.edu (Mike Clemens)
Subject: Re: midi file formats
Organization: Indiana University
Date: Fri, 2 Oct 1992 04:30:09 GMT

In article <EMUSIC-L%92100112011081@AUVM.AMERICAN.EDU> SEIFF@PACEVM.BITNET writes:
>Does anyone have a copy of the specifications for MIDI file formats,
>particularly Format 0 and Format 1. If so, could you please send it to me.
>Thank you.
>
>Joe Seif  <seiff@pacevm.bitnet>

I'd also like this if it's available... Thanks!

Mike Clemens
mclemens@silver.ucs.indiana.edu
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