=========================================================================
Date:         Sat, 29 Oct 1994 22:14:06 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         DAVID O'DONOGHUE <dodon@OZEMAIL.COM.AU>
Organization: OzEmail Pty Ltd
Subject:      Re[1]: Hi! What's This List Like?
 
Reply to message text:
 
>Hello Dan,
>welcome to the list from another Australian.  You'll also see David
>O'Donoghoe (spelling?),
 
Yes..I'm the son that Aaron Spelling never talks about - something to do with
my hygeine habits
(or lack thereof).
 
 
Regards
 
David O'Donoghoe - Spelling
DODON@OZEMAIL.COM.AU
=========================================================================
Date:         Sat, 29 Oct 1994 14:54:10 -0600
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         christine annette jackson <bambij@UNM.EDU>
Subject:      Re: Hi! What's This List Like?
In-Reply-To:  <m0r0m0s-000SrdC@eros.unm.edu>
 
On Fri, 28 Oct 1994, David Rodger wrote:
 
> Hello Dan,
> welcome to the list from another Australian.  You'll also see David
> O'Donoghoe (spelling?), Paul Copeland and Clive McFarlane, among others,
> who would seem to be Ozsters, too.  I mention these chaps because they're
> very vocal (or should that be literal?).
 
Welcome, Dan! (from an Albuquerquean who wishes she could at least visit
Australia:)...
 
I'm one of the non-vocal participants to this list because I so far have
only a pea-size bit of knowledge when it comes to e-music... But I
continue to read all the discussions in the hopes of some of it sinking
in! The list is great and these guys are very knowledgeable - worth sticking
around for!
 
Have fun!
 
Bambi
=========================================================================
Date:         Sat, 29 Oct 1994 18:46:00 -0400
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Jody Whitesides." <Amalgam@AOL.COM>
Subject:      Re: SMPTE and film frame rates
 
According to what I remember from film school (I graduated a few years ago),
the frame rate of 24 is what is considered standard, at least here in
American film.  I believe this is because this is the rate when film no
longer flickers during playback.  As far as when film is being shot:  24 fps
will get you the action as it happened, slower shooting speeds yield faster
motion on playback, and fast shooting speeds yield slower motion on playback.
 I don't recall the divisibility being the reason for the 24 frames.  I think
it has something to do with the measurement of the film itself.  I think each
frame is one inch in measure and is therefore measureable in our great
inches, feet, and yards, ect...  But as far as the rest of the world is
concerned I admit ignorance.
 
As far as video is concerned:  There are 30 frames a second with two fields
per frame, thus there are 60 fields a second.  This is because of the cycles
that our electricity spits out (a/c is 60 cycles a second), why, I don't
know.  Obviously this creates a transferance problem between the two media.
 So the engineers who make up the SMPTE body devised a couple of ways to may
them compatible.  There isn't one direct solution, so people choose which
they like best.  This could also be because American and European video
standards are not the same.  The 29.97 comes from the fact that 3 of the 60
fields-a-second are dropped each second during transferance.
 
I'm starting to babble so I'll let you go.  If there are more questions then
I will be happy to try and answer them.
 
Chuss!
 
Jody Whitesides
 
"It's what's played between the notes that's golden."
Amalgam@aol.com
=========================================================================
Date:         Sun, 30 Oct 1994 07:37:48 -0600
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Champagne Matthew G <mgc3295@USL.EDU>
Subject:      motion picture frame rate and flicker
 
24 frames per second is not a sufficient rate to eliminate
flicker from a mtion picture image, whichis why the
shutter in a projector interrupts the light twice each
frame, or 48 times each second. This yields a flicker-free
image.
 
By the way, the professional projectionist is a thing of the
past.
 
-Matthew Champagne, Master's candidate in music theory/comp
at USL and former part time projectionist. (mgc3295@usl.edu)
=========================================================================
Date:         Sun, 30 Oct 1994 17:22:44 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Michael P. Dougherty" <mdoughe@UNIX.CC.EMORY.EDU>
Subject:      A quick question
 
Hello,
 
I am new to the list, but have a question which I am hoping that some of
you e-musicians out there in e-land may be able to answer.
 
The question stems from a project in which I am currently involved on the
net, more specifically on the World Wide Web.  I am currently involved in
attempting to set up a node for music research which will, hopefully, be
able to open people around the world to music resources including library
catalogs, annotated bibliographies, and special music collections.  Of
course, music is a wonderful beast, because we need to be able to
transmit not only printed information, but also sound, and special visual
images, namely the music itself.  Thankfully, the first two of these is
taken care of.  However, other than scannable images which take up a
great deal of memory, it is difficult at the moment to send music
notation.  We have figured out that we can actually do this relatively
easily using several formats (ascii, and others) if we have a
non-licensed music font which we could make available to anyone on the
web.  This is where I am hoping that my new friends in e-land come in.
If you have any knowledge of such a beast, I would love to have any
information!!
 
Thanks
 
Michael P. Dougherty
=========================================================================
Date:         Sun, 30 Oct 1994 19:46:37 CST
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         James Porter <porterj@LBM.COM>
Subject:      SoundBlaster MIDI Troubles--HELP!!
 
A friend of mine is trying to use a SoundBlaster 16 card in his
486SX machine.  He is running Norton Desktop in Windows 3.1.  Each
time he runs Windows, he gets the following message:
 
Improper MIDIPORT setting in SYSTEM.INI
 
The SoundBlaster driver in Windows is set to MIDIPORT=330 (the only
other choice is 300).  I have ensured that the SYSTEM.INI file
matches this.
 
The machine will play MIDI files through the Media Player, but not
WAV files.
 
Anyone have any suggestions?  Please reply directly.
=========================================================================
Date:         Sun, 30 Oct 1994 18:55:08 -0800
Reply-To:     David Brewer <dbrewer@calvin.linfield.edu>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         David Brewer <dbrewer@CALVIN.LINFIELD.EDU>
Subject:      PAS 16 Midi Interface...
 
I am having a couple of problems with the 'built-in' midi interface on
the Pro Audio Spectrum 16 card.  First of all, whenever I use it I get a
lot of errors.  To be more specific, sometimes it sends rhythm
information but the wrong note information (so that it sounds like a kid
banging on a keyboard).  Another problem is that sometimes it will hit a
note and sustain it infinitely (so long that I eventually have to turn
off the keyboard to get the note to stop).
 
Those problems would be bad enough by themselves, but here's the REAL
problem.  When I use any MIDI software in Windows, the whole system locks
up within a few minutes.  It occurred to me that this might be an IRQ
problem, so I tried a few different settings.  On the default IRQ setting
(7), the digitized sounds and FM synthesis work fine but the external
MIDI is screwed up, and on IRQ 5 the external MIDI seems to work better
(still makes errors, but doesn't lock up) but the rest of the card goes
nuts (for example, WAV files cycle forever when played).  I tried many
other IRQ settings, but in each case I couldn't get both features to work.
 
This is driving me bonkers!  Does the PAS just have a naturally crappy
interface?  And do you think that the errors are most likely coming from
the interface or my equally cheapo keyboard? (Kawai FS780)
 
Any assistance would be greatly appreciated.
 
David Brewer
dbrewer@linfield.edu
=========================================================================
Date:         Mon, 31 Oct 1994 18:01:25 EST
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "<Clive McFarland>" <clive@PELICAN.DBE.CSIRO.AU>
Subject:      Re: A quick question
 
Michael P. Dougherty wrote:
 
[CHOMP]... it is difficult at the moment to send music
>notation.  We have figured out that we can actually do this relatively
>easily using several formats (ascii, and others) if we have a
>non-licensed music font which we could make available to anyone on the
>web.  This is where I am hoping that my new friends in e-land come in.
>If you have any knowledge of such a beast, I would love to have any
>information!!
>
Hi Michael
 
I'm not sure if this is what you're after, but the notation package Lime,
for instance, has screen and printer fonts (tufa and marl?) which are
installed (on a Mac in my case) and then allow full notation from what I
assume are ASCII-type files (I can check all this if necessary).  You may
well have already explored this possibility - they are provided free on the
demo version (available by ftp) but I don't know about any licensing
arrangement.  I can dig out the ftp site and author's email address (they
may even subscribe to this list...?) if you're interested email me.
 
Regards
 
Clive
 
Clive McFarland         CSIRO Australia         clive@pelican.dbe.csiro.au
=========================================================================
Date:         Mon, 31 Oct 1994 16:00:42 -0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Boris BASKEVITCH <Boris.Baskevitch@INFORMATIK.FH-WOLFENBUETTEL.DE>
Subject:      New midi (CAN)
 
Hi folks,
Do you remember, i told you about the CAN network for MIDI implementation,
i have a few more things to tell you about.
 
First, there was demand for my the report of my studies on the CAN. Actually,
i made a system for this network (a tranceiver for an implementation in German
trains), and to get my degree in electrical engeneering, i had to write a report
on it, it consists in an introduction on the CAN network, a description of the
protocol, and a description of the CAN chip that i used (registers...).
Some people asked me to put this report on the net, but i don't have access to a
FTP site for writing, does anybody here has an idea to help me (and them) ?
PS. This report was written with WinWord 6 and contains embedded diagrams.
 
I'm thinking of realising an interface between the MIDI and the CAN, just to
change the physical layer of it. I may use a microcontroller with the equivalent
of a CAN chip integrated on it, Motorola are making such a chip. If I succeed
with this, i will maybe, if i have some money left and enough knowledge for it,
make a kind of simple CAN soundcard for PCs that could simulate a simple
sequencer without using the actual MIDI network but directely the CAN network.
For this, i may need some help, is there anybody on this list who could give me
some documentation on hardware implementation of MIDI ?
If my system has a future, i could (maybe) be helped by a german company
specialised in the CAN network, but this is completely unofficial.
 
Then i would like to add a few things for the advantage of the CAN :
first the price, i had some information for my system, ie : a simple CAN chip
from Philips costs not more than 6 DM ($4.5), the same with a microcontroller
should not cost over $10.
 
Joe said :
>1) Many synths do *not* simply ignore data which is meaningless to them. In an
>   overload condition, in which many, many improperly formed messages are being
>   sent, many synths crash or hang, rather than just ignoring the overruns.
>   The fact that this can happen at all points to the bottleneck being in the
>   processing and not the propagation.
With a CAN chip, the controller will only listen to the messages adressed to it,
it will not be bothered at all by other, it is possible to program the CAN chip
to send an interruption only when a message with the corresponding ID is present
on the line. This is considerabely reducing the load to the controller.
 
>The classic question when tuning the performance of any computer network
>is, "Fix the worst response time first." In the case of MIDI, that's almost
>always going to be how fast the synthesizer processes events, rather than
>how fast events can flow.
Every CAN chip will send an ack bit on the line when a message has been
succefully recieved, if the ack is not present, the sender will repeat
the message. It is to the receiver controller to send a message to tell
the sender that the previous message has been succefully processeed, if it is
critical.
 
In my implementation of MIDI in the CAN, i will not change the protocol itself
it is too difficult for me at the moment, i will just adapt it to the CAN
protocol. But if a group of persons here feel able to improve it, i would be
glad to be part of it.
PS. i'm not a real musician, i only use MIDI for fun, i'm a french technical in
industrial computer programming and electronics, on the way to be a german
engineer on computing and automatism.
 
regards, Boris
Boris.Baskevitch@Informatik.FH-Wolfenbuettel.DE
------------------------------------------------------------------------------
=========================================================================
Date:         Mon, 31 Oct 1994 10:57:03 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Sanielevici Alexandre <sanielea@ERE.UMONTREAL.CA>
Subject:      duplicates
 
    Hi There.
I recently subscribed to this list, and keep recieving duplicate
copies of each message coming from this list. This morning I
recieved one in triplicate, and one in octuplate. What is causing
this and how can it be corrected (other than unsubscribing, of course!)
Thanks, Alex.
=========================================================================
Date:         Mon, 31 Oct 1994 17:54:13 GMT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Steven.Bramson@JET.UK
Subject:      MIDI speed
 
Joe McMahon wrote:
 
The thing I see as a point which may be obvious to everyone by now, but
what the heck:
 
The MIDI implementation has it own problems. However, the principle
bottleneck in MIDI networks is not usually the data transfer, but the data
response. Consider:
 
1) Many synths do *not* simply ignore data which is meaningless to them. In an
   overload condition, in which many, many improperly formed messages are being
   sent, many synths crash or hang, rather than just ignoring the overruns.
   The fact that this can happen at all points to the bottleneck being in the
   processing and not the propagation.
2) Let's make an estimate, which is probably not accurate, but should be in the
   ballpark: With an ability to deliver about 31 thousand bits per second,
   a maxed-out MIDI network should be able to deliver around thirty events per
   second on 16 channels (and this is assuming 1 wire, 8-byte messages). How
   many synths can process 30 events per second in close to real-time?
 
--
 
Here is a simple experiment.  Try playing a four or five note chord with each
hand (Am7 is easy and sounds OK).  Play the same chord with both hands (on
different octaves).  Now play that 8 or 10 note chord repeatedly as fast as you
can.  I reckon 4-6 times a second is possible, so up to 60 ons and offs per
second.  Now see if
 
a) your mother keyboard keeps up ie plays all the notes every time
b) your Mac/PC/Amiga/Atari MIDI can record that fast
c) any instruments down the line can keep up
 
I reckon there is probably something faster than MIDI between the mother
keyboard and its internal tone generators.
 
Steven.Bramson@jet.uk
 
PS on the subject of minor seventh cords, try playing Fm7 with your left hand
and Cm7 with your right hand together.  Then try Cm7 on the left and Fm7 on the
right.  Same notes but different octaves, yet to my ears, one sounds nice, the
other sounds a bit of a mess (I won't say which way round so as not to bias
your listening).  Any suggestions as to why?
=========================================================================
Date:         Mon, 31 Oct 1994 17:37:48 GMT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Steven.Bramson@JET.UK
Subject:      Re: PAS 16 Midi Interface
 
David Brewer writes
 
{tale of woe with intermittent faults and IRQ swapping omitted}
 
This is driving me bonkers!  Does the PAS just have a naturally crappy
interface?  And do you think that the errors are most likely coming from
the interface or my equally cheapo keyboard? (Kawai FS780)
 
--
 
Yes, I find similar problems with the PAS-16 MIDI.  Before I bought my PAS-16 I
bought an MPU-401 clone card.  This works perfectly with everything being a
hardware level compatible implementation rather than a UART mode only setup as
found on most general PC sound cards.  However I have had some problems with
Cubase Score if I have both interfaces plugged in.  I came to the coclusion some
time ago that there is no such thing as compatible.  I am beginning to think
that one should buy separate
 
- MIDI card
- External MIDI module for music generation
- WAV file play and record card
- SCSI card
 
rather than all in one almost but not quite compatible.  Saying that, the
MPU-401 uses the same default i/o address (330) as the Adaptec range of SCSI
cards.  So much for plug and play!
 
Steven.Bramson@jet.uk
=========================================================================
Date:         Mon, 31 Oct 1994 13:51:44 -0700
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Joe Grossman <grossmj@SPOT.COLORADO.EDU>
Subject:      Re: MIDI speed
In-Reply-To:  <01HIXEF1SA8G000CWQ@VAXF.COLORADO.EDU>
 
My guess would be, if you dislike the sound of the Cm7 in the bottom and
the Fm7 on top, is the minor 9th interval between the G and the A-flat.
                                                                Joe G.
=========================================================================
Date:         Mon, 31 Oct 1994 16:59:37 EST
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Jason Price <jason_price@CC.CRANBROOK.EDU>
Subject:      Audio VS. Digital
 
          To all Emusic-L subscribers:
 
          I am currently faced with the problem of buying new
          multi-track recording equip.  I am making a MAJOR
          investment, and don't want to blow a shit-load of money on
          someting that I will wish I hadn't done in a year or so.
          The two basic systems that I have come across that have
          really cought my attention are Tascam's DA-88 (8 track
          digital multitrack recorder) and an older Tascam
          Reel-to-Reel Analog 8 track Recorder.  The Reel-to-Reel is
          about $1,000 cheaper, but for that much, I can get a good
          two track recorder to master down to.  The feature about the
          DA-88 that I really like, is the fact that it has an
          internal card that records SMPTE time code and it doesn't
          have to waste a valuable track on your Multi-Track.  I have
          heard various things about both Digital and Audio equipment
          that make it so confusing to pick one that will provide
          great audio quality and won't become out-dated anytime soon.
           If anyone has any expierience or advise on which piece of
          equipment I should go for, I would like to hear it.
 
          THANX
 
 
          ------------------------------------------------------------
            Jason Price
            jason_price@cc.cranbrook.edu
            Bloomfield Hills, MI, USA
          ------------------------------------------------------------
=========================================================================
Date:         Mon, 31 Oct 1994 15:17:06 -0800
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Andre Smirnov <asmir@CCRMA.STANFORD.EDU>
Subject:      Russian Electroacoustic Music
 
Andre Smirnov
Director of the Theremin Center for Electroacoustic Music
at Moscow Conservatory of Music will present a LECTURE:
 
HISTORY AND PRESENT OF THE  RUSSIAN ELECTROACOUSTIC MUSIC
 
Wednesday, November 2 at 1:15 pm
 
in the Knoll Ballroom at the Center for Computer Research in Music
and Acoustics (CCRMA)
The Knoll, 660 Lomita Drive, Stanford University Campus
 
 
In the lecture  will be briefly mentioned some important but unknown
historical facts regarding the work of Russian inventors Leon
Theremin, Evgeny Murzin and Andrey Volodin in 20 - 60-s in
conjunction  with the recent situation in Russian electroacoustic
music and activities of the Theremin Center for Electroacoustic Music
with demonstration of video and music examples.
 
CONCERT OF RUSSIAN ELECTROACOUSTIC MUSIC
 
Thursday, November 3  at 4:00pm in the Knoll Ballroom.
 
1. Eduard Artemiev - I would like to returne  12:00
2. Albina Stoianova - Biriulki   8:30
 
3. Jorge Campos - Yaravi for flute, percussion and tape  11:30
 
Break
 
4. Aleksander Ponomariov/Valery Beluntsov/Andre Smirnov -
 
   Mrs. Lenine    15:00
Text-sound composition on the texts by Russian futurist poet
 
Velimir Khlebnikov "Gospozha Lenine" (1909-1912)
5. Vladimir Nikolaev - Echolalia for two fluts and tape   11:00
6. Vladimir Volkov - Portrait of Marina for flute and tape  12:00
 
for information and directions call 415 723-4971.
=========================================================================
Date:         Mon, 31 Oct 1994 19:04:03 -0500
Reply-To:     "L Mccoy (GD 1999)" <mcco5@minerva.cis.yale.edu>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "L Mccoy (GD 1999)" <mcco5@MINERVA.CIS.YALE.EDU>
Subject:      computer music conference
 
All subscribers:
 
 
I don't know if you have already been advised (since I joined the list
quite recently) but there is a conference entitled "Technology and the
Composer" which will be held at the University of Maryland (College Park)
from Nov.10-12.
 
Much good music.  Otto Laske will be there, as will Jean-Claude Risset
and Joji (sp?) Yuasa.  It should be a wonderful time to hear music and
discuss 'things'.
 
 
Hope to see some of you there!
 
Scott McCoy
mcco5@minerva.cis.yale.edu
=========================================================================
Date:         Mon, 31 Oct 1994 19:11:44 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Wesley Felteau <wfelteau@ST6000.SCT.EDU>
Subject:      Re: Audio VS. Digital
In-Reply-To:  <9410312239.AA34306@st6000.sct.edu> from "Jason Price" at Oct 31,
              94 04:59:37 pm
 
Get an Alesis ADAT or a Fostex "adat".
=========================================================================
Date:         Mon, 31 Oct 1994 23:31:21 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         john rossi <jrossi@BGNET.BGSU.EDU>
Subject:      Re: Audio VS. Digital
In-Reply-To:  <199410312339.SAA29913@falcon.bgsu.edu>
 
I believe that the most practical reason for choosing digital over analog
when it comes to 8-tracks is the size and the expense of the media.
People can argue till they are blue in the face about the relative
benefits of analog and digital and which sounds better, etc.  In reality,
however, the differences, if they exist in reality, are small.
Generally, I would have to believe that in order to compete with digital,
you would need at least a 1/2" 8-track deck and some kind of noise
reduction (dbx or Dolby s (or whatever the new Dolby is)).  Here is my
point.. A 40 min reel of decent analog 1/2" tape will cost you about
$30-$40 (with a hub) and it will occupy a box which is about 1'x 1' and
about 1" thick and which will weigh a couple pounds.  Contrast this to a
8mm tape which holds 100 min of audio, and is just slightly larger than
an audio cassette.  Even ADAT's S-VHS format is a hell of a lot handier
and less expensive than 1/2" analog.
 
John 3
=========================================================================
Date:         Sat, 22 Oct 1994 16:29:56 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         David Rodger <MUSDR@LURE.LATROBE.EDU.AU>
Subject:      Re: Abolish MIDI now
 
Hi folks,
 
I want a system which I can
send audio down  in real time and use  that as a modulation source for
my new hypothetical digital modular system. This isn't going to happen
until we go optical.
 So said Ab.
 
Does this mean sending this mod source audio along the same line as purely
control information?  If so, EVERYTHING will have to be sent with headers
to indicate what it is: control info, audio, sample data, whatever.  I
suppose that the status bytes of MIDI are like little headers which identify
the data to follow.  But then sysex and sample data is similarly headed, but
often occurs in longer datastreams, and not during real-time performance.
(Yes, I know there are sequencers out there which willl send sysex.  I'm
thinking more of sysex dumps here.)  If you wanted to retain real-time
performance inforamtion, then there would have to be an agreed way of
fragmenting the longer datastreams to maintain the timing of other types
of information.
 
Am I on the right track here?  More knowledgable people out there can correct
me or explain this better than me!
 
>My cheapo consummer
CD player has an optical output.
 
Yes, and most pundits out there thhink that TOSlink stinks (for audio and
other reasons), which probably explains why companies like Wadia prefer the
AT&T glass fiber interface for their gear.  In fact, even SP/DIF is better
on coaxial cable than TOSlink, and _that's_ over short runs.
 
>you  can  bet that  the very first
third  party    products to  emerge   will  be   <insert  name of  new
protocal>->midi converters
 
Yes.
 
>As for getting your old  gear to do all the  nice things that the  new
standard could do - tuff shit what do you expect.
 
I don't think anyone was suggesting that a new protocol would give old gear
new capabilities.  That isn't true even with plain ol' MIDI; remember the
MIDI implementation on the DX7s (xmit ch. 1 only, rx omni mode only) ?
I think people's concerns are for backwards compatibility, a point on which
you already offered a solution (third-party interfaces).
 
I think it's worth remembering that MIDI was made at a time when parallel
high-speed interfaces were more difficult and expensive to implement.  We
tend to forget how quickly things have progressed.  How many reading this
were using the internet ten years ago?
 
Regards, David.
=========================================================================
Date:         Sat, 22 Oct 1994 16:46:48 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         David Rodger <MUSDR@LURE.LATROBE.EDU.AU>
Subject:      Synth redundancy & sound in general
 
Hello everyone,
what an  interesting discussion this has been.  A few things occur to me.
 
When audio recordings didn't exist, or least weren't the most common way of
disributing musical ideas or works, musicians and composers relied on a
generally agreed standard for notating (making a record of) them.  This
implies that during that time, musical works were not properly finished
and would only be when performers reconstituted them in aural form.
 
Now, we can record (store, archive,...) completed works.  In fact, the
currency of audio recordings has been most raised by the popular music
industry, through which songs/pieces/works are intimately associated with
a particular artist/writer/producer.
 
Yet, the ease with which are able to do this also means that infinite versions
of the same ideas can also be propagated.  This is often encouraged; witness
the proliferation of records with multiple remixes and products like CD-ROMs
(such as Peter Gabriel's) which allow re-mixing of the song.
 
So, we have to ask ourselves, how closely _can_ we control the sounds through
which our ideas are expressed.  Well, I'll leave you all to ponder this and
also through another spanner in the works by suggesting that the answer has
always been "Not very well at all."  We have always been at the mercy of
different (inadequate or adequate) of signal processors, recoding technologies,
(sorry, 'recording') and reproduction systems which different people use.
That bandwidth-limited, dynamic range-limited, cheapie AM radio receiver with
a 3 inch speaker probably affects the sound more than a slight tweak on
the modulation amount on carrier 3 in  a DX7!
 
Thanks for the debate,
David.
=========================================================================
Date:         Sat, 22 Oct 1994 11:13:12 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         joe dierker <jade@VT.EDU>
Subject:      Re: Synth redundancy & sound in general
 
>
>Yet, the ease with which are able to do this also means that infinite versions
>of the same ideas can also be propagated.  This is often encouraged; witness
>the proliferation of records with multiple remixes and products like CD-ROMs
>(such as Peter Gabriel's) which allow re-mixing of the song.
>
>So, we have to ask ourselves, how closely _can_ we control the sounds through
>which our ideas are expressed.  Well, I'll leave you all to ponder this and
>also through another spanner in the works by suggesting that the answer has
>always been "Not very well at all."  We have always been at the mercy of
>different (inadequate or adequate) of signal processors, recoding technologies,
>(sorry, 'recording') and reproduction systems which different people use.
 
        For a well-anthologized if kinda old (though hardly square) view,
check out Walter Benjamin's "Art in the age of mechanical reproduction"...
[In _Illuminations_]...
 
 
                                                        Joe Dierker.
                                                        jade@vt.edu
=========================================================================
Date:         Sat, 22 Oct 1994 17:19:01 +0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Nick Rothwell <nick@CASSIEL.COM>
Subject:      Re: Nasty guitar sounds from a synth?
 
>(we have all signed a Just Say No To Chiffy Kalimba
>pledge for all future projects, you'll be happy to learn).
 
I didn't sign *nuffin'*.
 
 
    Nick Rothwell     CASSIEL Contemporary Music/Dance     nick@cassiel.com
=========================================================================
Date:         Sat, 22 Oct 1994 17:26:34 +0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Nick Rothwell <nick@CASSIEL.COM>
Subject:      Re: Timbral notation and extreme reactions to music
 
>(anyone remember the back covers of albums
>by M. Frog Labat, Tangerine Dream, and Klaus Schulze, tht depicted
>their bizarre notations on graph paper?).
 
I have BLACKDANCE and TIMEWIND which have Klaus's notation (in English,
interestingly). I use something similar, but I don't represent the sounds
graphically; instead I tend to use horizontal for time, vertical for pitch,
and will describe imagery in prose. The main reason for doing this is to
get some sense of temporal structure, some sense of orchestrational
development, and to make notes on MIDI assignments, effects settings, and
so on.
 
I didn't know that TD did anything like this; on the Ohr record sleeves
perhaps? One thing I did notice is that the latest (yawn) Virgin best-of
boxed set has excerpts called "Tangram (Puzzle Part)", "Rubycon (Dice
Part)", "Logos (Velvet Part)" and so on, which gives some insights into the
imagery, assuming the annotations are theirs to begin with.
 
>Which means that our music may in
>fact be the first to be rendered unrepeatable....
 
This discussion has come up before, and I think I was accused of some form
of elitism for not wanting to make the effort to make my music notated and
performable in some general sense. As Mike would say, "Feh."
 
>she claims, and I have
>independently proven in blind studies, that anything he's ever
>recorded instantly gives her a headache, regardless of playback
>volume.
 
I have one (1) authentic New Age album, a Californian solo-artist
contemporary piano thing which I have not managed to listen to right to the
end. It gives me a headache and makes me twitchy. I have no problem with
some quite heavy and noisy music (not to mention Schnittke), but this one
just puts me on edge...
 
 
    Nick Rothwell     CASSIEL Contemporary Music/Dance     nick@cassiel.com
=========================================================================
Date:         Sat, 22 Oct 1994 17:28:36 +0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Nick Rothwell <nick@CASSIEL.COM>
Subject:      Re: Live sequencing
 
>My
>approach has been to say that I know how much my brain and hands can
>do at any one time, and I know how much music is needed to fill out an
>arrangement that sounds good. If A < B, I am doing my audience a
>disservice by not providing them with the missing bits somehow.
 
True; I guess that my motivation is to strive for A=B somehow, by a
combination of reducing B (which is good for me: in the past I've been
tempted to over-orchestrate or over-sequence, and I'm after sparseness) and
increasing A:
 
>I
>suppose I could learn to play more and faster, and to redefine my
>style to allow for everything to be done live with two hands and two
>feet, but what will my audiences listen to while I'm relearning how to
>present my music?
 
Well, true, but it's possible to get there reasonably quickly with some
perseverance. Let me think ... For a gig in early 1993 I had Performer
doing some backing, with live stuff routed through MAX, but that included a
5-minute opening sequence which was just an interaction with MAX "tape
loops", and a five-minute closing sequence which was totally manual. By
December I'd managed a gig without a sequencer and without a keyboard, just
using MAX and the Peavey fader box.
 
The trick, of course, is to have some way of performing orchestrations,
either by making the orchestrations minimal, or by thinking a LOT about the
interactive systems. And of course, one still has to make some value
judgement about what counts as "sequenced" anyway (Wavestation wave
sequences? Arpeggios? Sampled drum rhythms?). But, for me anyway, I'm
convinced that this is the way to go: the creative tensions and discipline
feel right. It's still hard, though, and I'm not happy with the results so
far. Maybe in a year or so...
 
Oh yeah, another angle on this: what better way for finding ideas than to
improvise into orchestrational performance systems? I guess this is the
Vangelis philosophy.
 
 
    Nick Rothwell     CASSIEL Contemporary Music/Dance     nick@cassiel.com
=========================================================================
Date:         Sat, 22 Oct 1994 14:23:11 -0400
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         john rossi <jrossi@BGNET.BGSU.EDU>
Subject:      Re: Abolish MIDI now
In-Reply-To:  <199410212123.RAA06966@falcon.bgsu.edu>
 
Actually, the K2000s does have an optical interface as well as two
cabeled digital interfaces (SPDIF and whatever the pro one is called).
 
John 3
=========================================================================
Date:         Sat, 22 Oct 1994 14:29:34 -0400
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         john rossi <jrossi@BGNET.BGSU.EDU>
Subject:      Re: Nasty guitar sounds from a synth?
In-Reply-To:  <199410221623.MAA05966@falcon.bgsu.edu>
 
On Sat, 22 Oct 1994, Nick Rothwell wrote:
 
> >(we have all signed a Just Say No To Chiffy Kalimba
> >pledge for all future projects, you'll be happy to learn).
>
> I didn't sign *nuffin'*.
>
 
I rather like the sound of a well played Chiffy Kalimba, actually.
 
John 3
=========================================================================
Date:         Sat, 22 Oct 1994 14:59:50 -0400
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         john rossi <jrossi@BGNET.BGSU.EDU>
Subject:      Re: Timbral notation and extreme reactions to music
In-Reply-To:  <199410221638.MAA07062@falcon.bgsu.edu>
 
Back when I was playing with Pascal Languirand in the mid 70s, we learned
very early on that we needed some sort of notation for the synth parts so
that drummers and bass players would be able to cue.  Back then my synth
parts were quite etherial and were really more timbrally and
amplitudinally dynamic than they were melodic (March of the Plastic Ants
being an exception, BTW).  This, of course, was because all I had were
monophonic (duophonic in case of the EML-101) synths.  Also, my Synthi
AKS got most of the organizational work and anybody who has ever worked
with EMS equipment (especially with a sequencer not controlling pitch)
can realize how hard it would be to articulate the sound comming out in
order to explain where things go to a drummer.
 
We adopted a notation system which was similar to Klaus Schulze's but
used independent staves for frequency and timbre where the timbral
dynamics were drawn with continuous lines with beginnings on the staff at
points approximetly equal to the principal harmonic (i.e., filter
frequency and resonance point) and an opening and closing right faced
(i.e., <) acute angle extending and following the contour of the harmonic
content.  So, for example, a standard filter sweep would be anotated as a
blunted A-frame hut.  The degree of resonance was indicated by filling in
the 'triangle' with shades of grey.  For very high resonance settings the
'floor' would not be indicated just the vector of the filter frequency
(i.e., the roof of the hut) in a bolder ink line.  When superimposed on a
standard musical notation staff, it was easy to cue things to the
changing harmonic content.  For example, if a drummer was to start
playing at the beginning of a bar which occurred in time at a point about
two thirds through a filtered sweep of white noise then the point on the
drum notation would coincide with a point about 2/3 through the sweep
notation.
 
The only real problem was that Pascal was the only drummer we ever used
who could read this stuff.  Since he was generally playing a guitar or
synth it meant that one of us had to cue the drummer anyway (ambient
music is a real hard thing to count, if you think about it).  Actually,
it didn't matter since by looking at the notation we would know when
things were supposed to start and could cue them manually.
 
I think I still have some of the notation around from that period.  I'll
need to remember to look for it.
 
John 3
=========================================================================
Date:         Sun, 23 Oct 1994 10:02:17 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         DAVID O'DONOGHUE <dodon@OZEMAIL.COM.AU>
Organization: OzEmail Pty Ltd
Subject:      Re[1]: Synth Redundancy?
 
Reply to message text:
 
 
>It would probably be good enough for my purposes if I could get a sound
>on one synth that was reasonably close to one I had on another, but I
>don't consider myself a Synthesist, primarily. But I can imaginge that
>the many people on this list who spend hours upon hours tweaking a patch
>until it's just so are not going to be satisfied with 'reasonably close'
>or 'almost'.
 
I guess that if one wants to capture their music snap frozen with every nuance
and controller change intact, the only answer is to record to CD (or
Equivalent). But is change necessarily a bad thing? David Bowie has used many
different musicians to play his music in a live context, different musicians
indeed than played on the records..so we can expect the 'live' versions to
sound different from the record- does this make the music inferior? Perhaps
this is part of what makes live music an interesting experience- its
unpredictability.
 
I remember a piece of music I did that included a crappy little Casio keyboard
as one of the 'layers' in the main synth drone.
I have subsequently reworked it on other equipment, and BOTH versions have
their own appeal.
Yes, it IS unrealistic to expect that you can synthesize EXACTLY tha same
sounds on different equipment,
but thats OK! (IMHO)
 
Regards
 
David O'Donoghoe
DODON@OZEMAIL.COM
=========================================================================
Date:         Sat, 22 Oct 1994 10:50:58 +0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Oliver Scheel <os@HANNIBAL.LU.PFALZ.DE>
Organization: Private UUCP site, Ludwigshafen, FRG
Subject:      Re: Abolish MIDI now
 
>Isn't it about time that boring old serial MIDI is scrapped and replaced by
>a faster, parallel link with a wide data bus?
 
Perhaps. But the problem is not the MIDI speed. The synths are often too
slow to prozess the messages in time and if this isn't fixed before it
would be idiotic to create a new MIDI standard.
 
btw: I would prefer a serial link (it's much cheaper!) like ethernet. It's
fast enough to transmit simple MIDI data and even fast enough (1MBit/s) to
transfer simple audio data ...
 
Oliver
 
--
Oliver Scheel, Mundenheimer Str. 254, D-67061 Ludwigshafen (Germany)
phone: ++49 621 5296890 -- email: os@hannibal.lu.pfalz.de
=========================================================================
Date:         Sun, 23 Oct 1994 14:13:27 -0000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Boris BASKEVITCH <Boris.Baskevitch@INFORMATIK.FH-WOLFENBUETTEL.DE>
Subject:      Re: Abolish MIDI now
In-Reply-To:  Your message of "Fri,
              21 Oct 94 12:31:40 PDT."
              <199410230210.DAA19889@MAIL.FH-Wolfenbuettel.DE >
 
As a different bus for a new MIDI, i would like to propose the CAN.
The CAN (Controller Area Network) is is a message-oriented protocol
wich was designed to interconnect microcontrollers in car and industry
There are actually many supports for the data, (optical, ...) but a very
simple one a the line pair.
All the controllers are sychronised, there are no master/slave (multimaster
bus structure), that's why I thought of using it for a MIDI replacement.
The messages are basically composed of an ID (up to 2032) wich determines
the priority of the message (may be useful for MIDI), and the data itself
(0 to 8 bytes).
The transmission time is programmable (clock), the error detection/correction
is quite powerful, possibility of non-destructive bit-wise arbitration.
The CAN network follows the CSMA/CD protocol wich means more clearly that
if there is detection of a collision, the CAN chip takes in account the
priority of the messages and delays the messages.
 
I'm currently working on this network in Germany, so if anyone needs more
information on it, I will be glad to contact him/her.
 
PS. please excuse my english, I'm french, and I'm working in Germany, so I may
    mix a little bit.
 
Boris
 
------------------------------------------------------------------------------
Boris.Baskevitch@Informatik.FH-Wolfenbuettel.DE
------------------------------------------------------------------------------
=========================================================================
Date:         Sun, 23 Oct 1994 11:54:19 +0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Nick Rothwell <nick@CASSIEL.COM>
Subject:      Re: Abolish MIDI now
 
>Isn't it about time that boring old serial MIDI is scrapped and replaced by
>a faster, parallel link with a wide data bus?
 
Wide data bus? You obviously haven't seen the inside of my rack cases.
 
 
    Nick Rothwell     CASSIEL Contemporary Music/Dance     nick@cassiel.com
=========================================================================
Date:         Sun, 23 Oct 1994 11:55:24 +0000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Nick Rothwell <nick@CASSIEL.COM>
Subject:      Re: Abolish MIDI now
 
>Better yet, fiber.
 
Too fragile. (Have you seen what roadies do with cables? Have you ever
tried soldering fibre?)
 
>Maybe TCP/IP or
>something close.
 
Not realtime.
 
>(How would you like to be reading this on your synthesizer
>instead of your computer?  :-)
 
"A:88 MUL [IDONTTHINKSO]"
 
 
    Nick Rothwell     CASSIEL Contemporary Music/Dance     nick@cassiel.com
=========================================================================
Date:         Sun, 23 Oct 1994 22:53:31 MYT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Victor Voo <victor@SHINE.PC.MY>
Organization: Sunshine Audio & Video Productions
Subject:      Video - Shaping Your Sound
 
I live in Malaysia, a country literally on the other side of the
world, which makes 800 numbers inaccessible.  I would appreciate
it if someone could call First Light Video Publishing at:
 
    1-800-777-1576
 
and ask them for their FAX number and post it to me or this list.
I am interested to get their training video tape entitled
"SHAPING YOUR SOUND" at US$ 99.95 for use as a training tool for
my church sound system operators (about 20 of them).  Also, if
anyone has bought this set of videos, it would be great if you
could post your comments on it.
 
Thanks.
--
 =======================================================================
VICTOR VOO - Sunshine Audio & Video Productions :)
Fax   : (International Access Code)-60-3-9304611   Voice : 60-3-9301842
Email : victor@shine.pc.my
 =======================================================================
=========================================================================
Date:         Sun, 23 Oct 1994 13:38:11 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Keith C. Perry" <perrykc@DUNX1.OCS.DREXEL.EDU>
Subject:      Re: Abolish MIDI now
 
>> Isn't it about time that boring old serial MIDI is scrapped and replaced by
>> a faster, parallel link with a wide data bus?
>
>Better yet, fiber...
 
 
True, and that wouldn't be hard to do :)  Someone here are Drexel is
working on that... :)
 
 
 
===+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+===
                                              \/
Keith C. Perry 800-836-4599                   /\
       ______________________                 \/ "To be or not to be...
Owner, >Studio K Productions<                 /\
       ^^^^^^^^^^^^^^^^^^^^^^                 \/     THAT AIN'T A QUESTION"
                &                             /\
Floor Supervisor / Senior Consultant          \/
Drexel University Office of Computing Services--|||||||||||||||||||||||||||||
...
=========================================================================
Date:         Mon, 24 Oct 1994 13:06:35 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         DAVID O'DONOGHUE <dodon@OZEMAIL.COM.AU>
Organization: OzEmail Pty Ltd
Subject:      Re[1]: Synth Redundancy?
 
Reply to message text:
 
 
>It's not so  much  tweaking a patch  so  that  it's just so   but more
>setting it up so that it responds in certain (often very bizarre) ways
>to performance  controls.  Thats   the real   problem with  trying  to
>recreate a lost sound jst from the midi track.
 
>        Ab.
 
Absolutely.
I remember assigning both pitch (with a two octave range) and transwave loop
point to the modulation wheel of a VFX-SD.
I wouldnt like to try and recreate the same sound on my JV-80! But thats OK, If
I decided to rework that piece of music I would do another sound which doesnt
sound quite the same, or respond to the controller message in the same way, but
would sound quite cute in its own right- and appropriate to the piece.
 
Regards
 
DAVID O'DONOGHOE
DODON@OZEMAIL.COM.AU
=========================================================================
Date:         Mon, 24 Oct 1994 13:06:41 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         DAVID O'DONOGHUE <dodon@OZEMAIL.COM.AU>
Organization: OzEmail Pty Ltd
Subject:      Re: Synth Redundancy?
 
Reply to message text:
 
 
> remixing the same sequence with other machines leads to odd results.
 
'Odd' can be good!
 
>. He sent effects settings, a SMF, even patch data and
>a tape of his mix; we didn't get anywhere near it in the end.
 
Goes to show that no matter how much experience we've had, and/or how clever we
are,
we are STILL at the mercy of our machines to a point (amongst other dark forces
such as our current emotional state, which can ALSO be influenced by our
machines!)
 
 
DAVID O"DONOGHOE
DODON@OZEMAIL.COM.AU
=========================================================================
Date:         Sun, 23 Oct 1994 20:07:10 -0700
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         atomic city <atomic@NETCOM.COM>
Subject:      Re: Nasty guitar sounds from a synth?
In-Reply-To:  <199410221826.LAA02225@mail.netcom.com> from "john rossi" at Oct
              22, 94 02:29:34 pm
 
john rossi writes--
>
>On Sat, 22 Oct 1994, Nick Rothwell wrote:
>
>> >(we have all signed a Just Say No To Chiffy Kalimba
>> >pledge for all future projects, you'll be happy to learn).
>>
>> I didn't sign *nuffin'*.
>>
>
>I rather like the sound of a well played Chiffy Kalimba, actually.
>
>John 3
 
Mutiny! MUTINY! Tie the scurvy dogs to the yardarm! Arrrr! :)
 
 
--
mike metlay * atomic city * box 81175 pgh pa 15217-0675 * atomic@netcom.com
---------------------------------------------------------------------------
"I've had some beautiful women melt my heart...Now a Chroma is melting it."
                                                                  (r. fahl)
=========================================================================
Date:         Mon, 24 Oct 1994 14:46:25 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         David Rodger <MUSDR@LURE.LATROBE.EDU.AU>
Subject:      Controller Area Network
 
Hi folks,
recently, Boris wrote:
>The CAN (Controller Area Netwrok is a message-oriented protocol
>which was designed to interconnect microcontrollers...
>All the controllers are synchronised, there are no master/slave (multimaster
bus structure)...
 
So, if you're wanting to use this for MIDI-like communications, how do you
implement any kind of master/slave relationship, e.g. synth responding to
sequencer's messages?
 
>The messages are basically composed of an ID (upto 2032)...
 
Odd number.  I guess this is how you do the master/slave thang.  Sequencer
sends messages of ID nnnn and synth is programmed to respond to messages of
ID nnnn.
 
Sounds fascinating.  Tell us more, please!
 
Regards, David.
<musdr@lure.latrobe.edu.au>
=========================================================================
Date:         Mon, 24 Oct 1994 12:37:32 +0200
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "A. Fiorletta" <ccd505@CCD.UTOVRM.IT>
Subject:      chord recognition
 
I'm doing a software real time arranger...
Does any one know about algoritms for chord recognition (how they work),
or can tell me where i can read something about?
Thank you, Alessandro Fiorletta
=========================================================================
Date:         Mon, 24 Oct 1994 09:52:37 CDT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Kevan L. Moore" <kevan@MX6.MSFC.NASA.GOV>
Subject:      BOSE acoustic wave
 
Hello all.
 
        Anybody out there using a BOSE acoustic wave for
        their equipment?
 
        How well does it work?
        Any problems?
        Is it LOUD enough?
        Is it portable enough?
 
any comments welcome.
 
 
Thanks.
 
kevan
=========================================================================
Date:         Mon, 24 Oct 1994 17:09:45 -0000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Boris BASKEVITCH <Boris.Baskevitch@INFORMATIK.FH-WOLFENBUETTEL.DE>
Subject:      Re: Controller Area Network
In-Reply-To:  Your message of "Mon,
              24 Oct 94 14:46:25 +1000."
              <199410240448.FAA04143@MAIL.FH-Wolfenbuettel.DE >
 
David asked how to implement a master/slave relationship with the CAN,
This kind of relationship is implemented by the message identifieres (as
you guessed). For example, a sequencer (with the highest priority) is Id'd
number 0, it sends a request (a note) to an expander ID'd number 1 (lower
priority). The other controllers (synth...) may or may not listen to the
message, depending if they are ID'd number 1 or not.
 
I would like to add that each message has a variable lengh(t) of 0 (just for
synchronisation) to 8 byte (that could code any kind of modulation of the
note)
 
I remember that somebody said that synth. would not be intelligent enough
to control such a network, as an answer, I would like to say that the CAN
network is supported by single chips (from Philips or Intel) that handle
the whole protocol.
 
Boris
 
------------------------------------------------------------------------------
Boris Baskevitch,   for a new MIDI
Boris.Baskevitch@Informatik.FH-Wolfenbuettel.DE
------------------------------------------------------------------------------
=========================================================================
Date:         Mon, 24 Oct 1994 08:05:04 -0700
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         atomic city <atomic@NETCOM.COM>
Subject:      Re: Timbral notation and extreme reactions to music
In-Reply-To:  <199410221635.JAA17165@mail.netcom.com> from "Nick Rothwell" at
              Oct 22, 94 05:26:34 pm
 
Nick Rothwell writes--
>
>>(anyone remember the back covers of albums
>>by M. Frog Labat, Tangerine Dream, and Klaus Schulze, tht depicted
>>their bizarre notations on graph paper?).
>
>I have BLACKDANCE and TIMEWIND which have Klaus's notation (in English,
>interestingly). I use something similar, but I don't represent the sounds
>graphically; instead I tend to use horizontal for time, vertical for pitch,
>and will describe imagery in prose. The main reason for doing this is to
>get some sense of temporal structure, some sense of orchestrational
>development, and to make notes on MIDI assignments, effects settings, and
>so on.
 
I used to do very intricate drawings on graph paper, with lines denoting
timeflow and breakpoints and shadings and curves for timbre. Now that I
do most work on a sequencer, I find its graphic abilities overshadowing
mine and changing how I work.
 
>I didn't know that TD did anything like this; on the Ohr record sleeves
>perhaps?
 
Yep. When you come visiting next, remind me to show you the inner gatefold
sleeve of "alpha centauri".
 
>One thing I did notice is that the latest (yawn) Virgin best-of
>boxed set has excerpts called "Tangram (Puzzle Part)", "Rubycon (Dice
>Part)", "Logos (Velvet Part)" and so on, which gives some insights into the
>imagery, assuming the annotations are theirs to begin with.
 
Big assumption.
 
>>Which means that our music may in
>>fact be the first to be rendered unrepeatable....
>
>This discussion has come up before, and I think I was accused of some form
>of elitism for not wanting to make the effort to make my music notated and
>performable in some general sense. As Mike would say, "Feh."
 
Me? Never! Libel! Slander!
 
>I have one (1) authentic New Age album, a Californian solo-artist
>contemporary piano thing which I have not managed to listen to right to the
>end. It gives me a headache and makes me twitchy. I have no problem with
>some quite heavy and noisy music (not to mention Schnittke), but this one
>just puts me on edge...
 
Feh! Oops.
 
 
--
mike metlay * atomic city * box 81175 pgh pa 15217-0675 * atomic@netcom.com
---------------------------------------------------------------------------
"How does it sound? Well, if jackhammers could sing...." (d. a. c. crowell)
=========================================================================
Date:         Mon, 24 Oct 1994 10:10:44 CDT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Bob Crispen <crispen@EIGHT-BALL.HV.BOEING.COM>
Subject:      Re: Abolish MIDI now
 
Arne Claassen ISE <claassen@EBS330.EB.UAH.EDU> sez:
 
>The problem with something like TCP/IP is that it is a packet system, while
>MIDI requires real-time continous data flow.
 
Let me expand on that for a minute.  The trouble with TCP/IP is that
(a) it's the wrong level of protocol, and (b) it tells you stuff you
don't want to know.
 
To take the second first, TCP/IP requires a handshake between the
parties to a virtual connection between machines.  Before the
originating machine can send the next packet, the receiving machine
has got to acknowledge receipt of the packet.  After a period of
time (always at least several seconds) if a packet isn't acknowledged,
it's re-sent.
 
Now think about that for a minute.  Let's say your receiving machine
misses a note-on.  If you were using TCP/IP, no problem -- you'll
probably get that note-on 10 seconds later!
 
The point is, if you get back an error message, or failure to acknowledge
a message, what can you do with it that won't make things worse?
 
What can you do with the concept of a virtual connection that won't
make things worse?  Well, perhaps you can have module B respond if
module A is shut down.  That's kinda cool.  But if you have a module
that only plays on the last song of a 2-hour set, would you want to
have its connection time out?  How would you prevent that?
 
Worse, every non-functional message such as an acknowledgement is a
message just the same as a note-on command message when you think about
bus contention.  If you think it's a pain having a live keyboard and a
sequencer talking to the same MIDI module on the same channel now, just
imagine what it would be like with every MIDI module sending back an
acknowledgement to a sequencer for every note!
 
Finally, MIDI is at *highest* a MAC-level (Media Access Control
level) protocol.  And that's just the right level.  MIDI doesn't have
the problems of managing error correction because it doesn't *have*
error correction.  It doesn't have the problem of managing connections,
routing, or level of service because the standard doesn't know about
and doesn't care about those things.
 
As a result, it's cheap.  A $29.95 toy can have some sort of MIDI
interface.  As a purchaser of FDDI boards from time to time, I know
that a more complicated substitute for MIDI would add hundreds of
dollars to the price of every synth, effects module, and MMC device.
And that's liable to stay the case for a long time, since there aren't
enough musicians to underwrite the expense of the development of the
MIDI equivalent of the Protocol Engine or even the LANCE chip.
+-------------------------------+--------------------------------------+
| Rev. Bob "Bob" Crispen        | Music should not be held responsible |
| crispen@foxy.hv.boeing.com    |   for the people who listen to it.   |
+-------------------------------+--------------------------------------+
=========================================================================
Date:         Mon, 24 Oct 1994 08:09:13 -0700
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         atomic city <atomic@NETCOM.COM>
Subject:      Re: Timbral notation and extreme reactions to music
In-Reply-To:  <199410221857.LAA06501@mail.netcom.com> from "john rossi" at Oct
              22, 94 02:59:50 pm
 
john rossi writes--
>
>Back when I was playing with Pascal Languirand in the mid 70s, we learned
>very early on that we needed some sort of notation for the synth parts so
>[...really nifty description deleted...]
>I think I still have some of the notation around from that period.  I'll
>need to remember to look for it.
 
We all ought to scan this stuff and upload it to the archives so people
can see examples of this sort of work; maybe bootleg the Schulze and
TD art, some Stockhausen graphic scores etc.? I mean, there's an art
gallery of old synths on Analog Heaven's Web page....
 
--
mike metlay * atomic city * box 81175 pgh pa 15217-0675 * atomic@netcom.com
---------------------------------------------------------------------------
"How does it sound? Well, if jackhammers could sing...." (d. a. c. crowell)
=========================================================================
Date:         Mon, 24 Oct 1994 13:15:32 +0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Stefan Scheffler <stefan@MMS-GMBH.DE>
Subject:      Re: Abolish MIDI now
 
>The problem with something like TCP/IP is that it is a packet system, while
>MIDI requires real-time continous data flow. I'd suggested SCSI a year
>ago but got shot down for the same reason, in addition that it only has
>7 nodes...
Think about its successor FireWire, it will implement beside stream
channels realtime capable channels. As far as I know it will support
several billion devices on the same line :-). Since it will come very soon
it might be a good opportunity to jump on this for a new MIDI.
 
Ciao,
  Stefan
 
<> stefan@mms-gmbh.de
<> Stefan Scheffler, Software Development
<> Steinberg Soft- & Hardware GmbH
<> Cubase Lite/Score/Audio for Mac/Atari/Windows
=========================================================================
Date:         Mon, 24 Oct 1994 12:41:52 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Macalester College Music Department <FORNER@MACALSTR.EDU>
Subject:      Re: Nasty guitar sounds from a synth?
In-Reply-To:  Your message dated "Sat,
              22 Oct 1994 14:29:34 -0400" <01HIKUD0LHWYAFVLDI@MACALSTR.EDU>
 
> On Sat, 22 Oct 1994, Nick Rothwell wrote:
>
> > >(we have all signed a Just Say No To Chiffy Kalimba
> > >pledge for all future projects, you'll be happy to learn).
> >
> > I didn't sign *nuffin'*.
> >
>
> I rather like the sound of a well played Chiffy Kalimba, actually.
>
> John 3
 
Particularly in B Flat minor
 
ef
=========================================================================
Date:         Sun, 23 Oct 1994 10:11:07 PM
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Stephen Travis Pope <stp@CNMAT.CNMAT.BERKELEY.EDU>
Subject:      Re: Cypher availability
In-Reply-To:  elad ophir <eo536318@ccsf.cc.ca.us>'s letter of: 94 Oct 21
 
The LISP source code for Cypher is included on a CD-ROM with the book
"Interactive Music Systems" by Robert Rowe, MIT Press, 1992.
 
stp
 
___Stephen Travis Pope
___Editor, Computer Music Journal, MIT Press
___Center for New Music and Audio Technology
___Dept. of Music, U. C. Berkeley
___stp@CNMAT.Berkeley.edu, (+1-510) 644-3881
=========================================================================
Date:         Fri, 21 Oct 1994 23:48:30 +6816
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Steven.Bramson@JET.UK
Subject:      Re: Abolish MIDI now
 
Well it is a while since we had the last interesting discussion on this
topic.  I recall extravagant claims for firewire, although a few months ago I
saw a learned article about the technology which looks very interesting.
Here are my thoughts:
 
Obvious problems
 
- 16 channels is too few and can be sucked up by one expander, so you need
  to change the protocol
- MIDI sample damps are slow, so you need to use uprated hardware
 
Use ethernet.
 
- Both interface and cables are cheap and widely available.
- The cable is reasonably robust.
- You can go a long way (twice round a large stage).
- 10 Mbits is 300 times faster than 30 kbits.
- It may not be a real time point to point link, but that shouldn't matter
  if it is fast enough, and it is.
 
Addressing
 
- Every synth gets a unique ethernet address.
- Allow 65,536 channels per address.
- Instruments broadcast to all addresses.
- You programme each synth as to whether or not to listen.
- By default, instruments only listen to 16 channels (backward compatible
  with MIDI).
 
Protocol
 
- New types of message which do interesting things go as a new
  non-manufacturer specific sysex.  This stuff can just be ignored
  or filtered for backward compatibility.
 
- Some new features:
  independent note pitch bend
  allowing a long note to change volume in mid note
  extended pitch range and precision
  extended volume range and precision
 
Any more suggestions or flames?
 
Steven.Bramson@jet.uk
 
PS, nice to see JR3 back with us.
=========================================================================
Date:         Mon, 24 Oct 1994 22:34:23 -0000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Boris BASKEVITCH <Boris.Baskevitch@INFORMATIK.FH-WOLFENBUETTEL.DE>
Subject:      Re: Abolish MIDI now
In-Reply-To:  Your message of "Fri,
              21 Oct 94 23:48:30."
              <199410241929.UAA11351@MAIL.FH-Wolfenbuettel.DE >
 
Steven says about ethernet for a new MIDI :
>Use ethernet.
 
The high baud rate sounds very interesting, accually, the whole thing
looks like the CAN.
You say :
>- You programme each synth as to whether or not to listen.
but on ethernet, i think that each tranceiver as an adress and sends
to adress(es), is it possible to broadcast ? What about synchronisation ?
 
>- By default, instruments only listen to 16 channels (backward compatible
>  with MIDI).
What kind of compatibility ?
=========================================================================
Date:         Mon, 24 Oct 1994 16:25:37 -0600
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         christine annette jackson <bambij@UNM.EDU>
Subject:      Re: Video - Shaping Your Sound
In-Reply-To:  <m0qz7So-000SoxC@eros.unm.edu>
 
On Sun, 23 Oct 1994, Victor Voo wrote:
 
> I live in Malaysia, a country literally on the other side of the
> world, which makes 800 numbers inaccessible.  I would appreciate
> it if someone could call First Light Video Publishing at:
>
>     1-800-777-1576
>
> and ask them for their FAX number and post it to me or this list.
 
Hey, Victor!
 
The FAX # is (310) 558-7891.
 
Good luck!
 
Bambi
=========================================================================
Date:         Tue, 25 Oct 1994 18:39:16 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         DAVID O'DONOGHUE <dodon@OZEMAIL.COM.AU>
Subject:      Re: Abolish MIDI now
 
Reply to message text:
 
 
>- Some new features:
>  independent note pitch bend
 
That would be nice
 
>  allowing a long note to change volume in mid note
 
Controller 7 already does this (doesnt it?)
 
Regards
 
DAVID  O"DONOGHOE
DODON@OZEMAIL.COM.AU
=========================================================================
Date:         Tue, 25 Oct 1994 06:18:04 -0700
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         atomic city <atomic@NETCOM.COM>
Subject:      From the Management: Please Read
 
No sarcastic *sigh*s, please, folks. While there are a number of
people out there who are simply too lazy to remember how to
unsubscribe themselves properly, there are others who try and for some
reason cannot do it. There are usually reproducible reasons for this;
the one that comes immediately to mind is that a subscriber's local
mail router, hence their address, changes, and the list server doesn't
know about it. If an SIGNOFF message is sent from the new address, the
List Server will ignore it because it doesn't match the address on file.
This is one problem that Joe and I don't mind fixing, although it may
take us a day or two to process it by hand and we may have to talk to
you if the information you provide us is not complete or correct.
 
If you are having trouble unsubscribing, do NOT post a general help
call to the list; send private email to me AND Joe (atomic@netcom.com,
xrjdm@farside.gsfc.nasa.gov) with your current and old email
addresses. If the List Server's really wedged, we may have to go up
one rung on the AUVM hierarchy to get it fixed, so be patient. This
service is for those with real problems (i.e. tried at least twice to
unsubscribe properly, but couldn't; remember the command to send to
the list server, listserv@american.edu, is 'SIGNOFF <listname>' with
no username). Many times this problem can be avoided simply by
unsubscribing yourself temporarily before the switch is made and
resubscribing afterwards; it's rare that a switch is performed without
any prior warning to users.
 
One other note. We have been informed that the List Server and all of
the lists are being updated soon to a pure-Internet protocol; the
Bitnet address <list>@AUVM and the Internet address
<list>@auvm.american.edu will be superseded by <list>@american.edu.
You should change your aliases now. Anyone who is completely restricted
to BITNET-based access and is utterly incapable of Internet use should
contact us, so we can get a feel for how to continue your service.
 
Virtually yours,
mike
for the EMUSIC-L and SYNTH-L management
 
--
mike metlay * atomic city * box 81175 pgh pa 15217-0675 * atomic@netcom.com
---------------------------------------------------------------------------
"How does it sound? Well, if jackhammers could sing...." (d. a. c. crowell)
=========================================================================
Date:         Tue, 25 Oct 1994 14:37:56 CET
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Jacek Latanowicz <JACK%PLPUAM11.BITNET@SEGATE.SUNET.SE>
Subject:      Re: From the Management: Please Read
In-Reply-To:  Message of Tue,
              25 Oct 1994 06:18:04 -0700 from <atomic@NETCOM.COM>
 
On Tue, 25 Oct 1994 06:18:04 -0700 atomic city said:
>No sarcastic *sigh*s, please, folks. While there are a number of
*sigh* :)
 
>
>Virtually yours,
Mike, My virtual GF writes me this kind of sentences :) hehehe
 
Jacek L.
 
 
>mike
>for the EMUSIC-L and SYNTH-L management
>
>--
>mike metlay * atomic city * box 81175 pgh pa 15217-0675 * atomic@netcom.com
>---------------------------------------------------------------------------
>"How does it sound? Well, if jackhammers could sing...." (d. a. c. crowell)
=========================================================================
Date:         Tue, 25 Oct 1994 06:55:26 -0700
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         atomic city <atomic@NETCOM.COM>
Subject:      Re: Abolish MIDI now
In-Reply-To:  <199410250836.BAA25096@mail.netcom.com> from "DAVID O'DONOGHUE"
              at Oct 25, 94 06:39:16 pm
 
>>  allowing a long note to change volume in mid note
>
>Controller 7 already does this (doesnt it?)
 
Not necessarily. Knowing what I know about how some synths are
manufactured, it wouldn't surprise me if some synth architectures
(especially on older or cheaper digital synths) are set up so that the
synth itself samples volume at note-on and holds it there for as long
as the note is sounding. It saves processor load inside the synth, but
removes a level of dynamic control. If true (note that I'm making an
educated guess here, and cannot cite examples), this is a failing of
the synth, not of MIDI. As for individual-note pitch bends, this is
not a feature of MIDI 1.0, true, but how would you implement it? Most
folks who HAVE to have realtime per-note articulations within the
current standard simply use polyphonic pressure. The resolution is
poorer, but it's in there now, it works, and is supported by many
synths, at least as received data if not transmitted (you can paint it
in with a sequencer).
 
\newenvironment{Rant}[1]{\em #1\/}
 
\begin{Rant}{This is all IMO, of course, but....}
 
Personally, I believe that MIDI isn't the heart of the problem at all;
it's just a handy scapegoat. While there are some things that are
real, noticeable, and definitely MIDI's fault (lack of resolution in
Control Changes and/or sloppy handling of microtonality, for example),
many others are either the fault of the synths themselves, or even
mythical. I'm not too young to remember how MI industry people quoted
a three-millisecond-per-link cascading delay of MIDI through synths as
gospel for years, until in a controversial guest editorial in KEYBOARD
the head of MIDIMIX (nice company. I miss them; are they still in
business?) categorically called it a lie and supported his claim with
facts. And as a happy but occasionally exasperated owner of a Roland
U-50, I know for a fact that a lot of synth manufacturers aren't
putting near enough horsepower in their boxes. What good is a new
'standard' for synths that can't hack the old one? The only faint ray
of hope I see has nothing to do with ZIPI or CAN or any other cute
acronym; it's the (probably meaningless) claim by Roland that their
newest tone-box has a high-speed RISC processor in that can handle all
I/O and sound generation options at a reasonable throughput rate.
 
I regard these various new standards for control as worthwhile araes
for research, and pointers to the next generation of music control
(when it comes). But until we're all making music on 200MHz RISC audio
workstations over fiber-optic LANs, don't try to make me believe we
need them right now for the machines we have. If anything, the market
is being driven toward cheaper and simpler machines thanks to General
MIDI and computercarditis, not toward more complex datamunchers. Nice
cheap reliable easy-to-implement two-optoisolators-and-a-DIN-jack MIDI
is not getting less relevant to the music industry, but MORE relevant.
 
My studio is controlled by a reasonably inexpensive Mac with tightly
integrated software and a powerful MIDI interface with its own CPU
(anyone with more than three synths and/or two keyboard controllers
and/or a MIDI data processing device with merging and throughput like
an arpeggiator or an MRC who thinks a Studio 5LX is a waste of money
should have his head examined), and I know damn well where things are
bogging down and it ain't MIDI. Discussing new improvements to MIDI is
fine, but a clarion call to launch a new industry standard when the
one we have is underutilized and growing more popular and widespread
in terms of market share every day is rather quixotic and is bound to
lead to indigestion.
 
\end{rant}
 
Oo. I haven't written one of those in a WHILE! Kinda feels like having
your back scratched, doesn't it? ;->
 
mike
--
mike metlay * atomic city * box 81175 pgh pa 15217-0675 * atomic@netcom.com
---------------------------------------------------------------------------
"How does it sound? Well, if jackhammers could sing...." (d. a. c. crowell)
=========================================================================
Date:         Tue, 25 Oct 1994 19:40:38 +0000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Nick Rothwell <nick@CASSIEL.COM>
Subject:      Re: Abolish MIDI now
 
>Think about its successor FireWire,
>....
>Since it will come very soon
 
Ah yes, who was it who told me that a year ago...?
 
 
    Nick Rothwell     CASSIEL Contemporary Music/Dance     nick@cassiel.com
=========================================================================
Date:         Tue, 25 Oct 1994 19:42:49 +0000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Nick Rothwell <nick@CASSIEL.COM>
Subject:      Re: Timbral notation and extreme reactions to music
 
>Now that I
>do most work on a sequencer, I find its graphic abilities overshadowing
>mine and changing how I work.
 
Danger Will Robinson. I found that as well, but also found that I was
getting hung up on things because I couldn't scribble quick notes; if the
layout was in the sequencer I quickly got tied down in "implementation"
details and stuff like that. Now I still do stuff on screen, but the paper
is here as well because that acts as a cache for implementation notes which
I can scribble on quickly without losing any creative process I'm working
on.
 
As it happens, I'm back using a sequencer (Vision) for the first time in
nearly a year. I'm not actually recording MIDI into it yet, just using it
as a sort-of cue sheet for now; MAX does most of the work (via beat sync)
and that in turn drives the R-8.
 
>Yep. When you come visiting next, remind me to show you the inner gatefold
>sleeve of "alpha centauri".
 
Oooh.
 
>Feh! Oops.
 
*Ding!*
 
 
    Nick Rothwell     CASSIEL Contemporary Music/Dance     nick@cassiel.com
=========================================================================
Date:         Tue, 25 Oct 1994 16:00:51 EST
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Jason Price <jason_price@CC.CRANBROOK.EDU>
Subject:      Ethernet???
 
          Can someone help me out here, I am fairly new the EMusic-L
          mailing list, and I don'st understand what this Ethernet
          thing is.  Is this a network that you can sign on with your
          computer that broadcasts MIDI Signals????  I'm confused!
 
 
          jason_price@cc.cranbrook.edu
=========================================================================
Date:         Wed, 26 Oct 1994 07:36:46 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         DAVID O'DONOGHUE <dodon@OZEMAIL.COM.AU>
Subject:      Re: Abolish MIDI now
 
Reply to message text:
 
>>>  allowing a long note to change volume in mid note
>>Controller 7 already does this (doesnt it?)
 
>Not necessarily. Knowing what I know about how some synths are
>manufactured, it wouldn't surprise me if some synth architectures
>(especially on older or cheaper digital synths) are set up so that the
>synth itself samples volume at note-on and holds it there for as long
>as the note is sounding
 
True..(try it on a Poly 800) most modern synths are OK in this regard, however.
 
> If true , this is a failing of the synth, not of MIDI.
 
Exactly
 
>As for individual-note pitch bends, this is not a feature of MIDI 1.0, true,
> but how would you implement it? Most
>folks who HAVE to have realtime per-note articulations within the
>current standard simply use polyphonic pressure.
> The resolution is poorer, but it's in there now, it works, and is supported
by many
>synths, at least as received data if not transmitted (you can paint it in with
a sequencer).
 
Again, it comes down to the limits imposed on us by the SYNTH.
As you noted, not EVERY synth transmits poly pressure (important to the sort of
person who wants pitchbend per note). Not every synth RECEIVES poly pressure.
And of those that DO receive it, not all of them will allow you to route PITCH
as the thing to be modulated.
Ho Hum..Problems, Problems...
 
> it's the (probably meaningless) claim by Roland that their
>newest tone-box has a high-speed RISC processor in that can handle all
>I/O and sound generation options at a reasonable throughput rate.
 
I HOPE it means that Continuous controllers will be handled ALOT faster.
I should be getting one at the shop this week...I guess we'll find out exactly
HOW MUCH this RISC chip means very soon....
 
Regards
 
DAVID O"DONOGHOE
DODON@OZEMAIL.COM.AU
=========================================================================
Date:         Tue, 25 Oct 1994 16:54:55 CDT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Kevan L. Moore" <kevan@MX6.MSFC.NASA.GOV>
Subject:      polyphonic pitch-bending
 
> >As for individual-note pitch bends, this is not a feature of MIDI 1.0, true,
> > but how would you implement it? Most
> >folks who HAVE to have realtime per-note articulations within the
> >current standard simply use polyphonic pressure.
> > The resolution is poorer, but it's in there now, it works, and is supported
> by many
> >synths, at least as received data if not transmitted (you can paint it in
 with
> a sequencer).
 
how about a note on event with a 2-byte pitch instead of a note number?
using a scheme like that any pitch from 0 - 65535 hz could be specified.
unless you get WAY down into the bass range, 1 hz won't make a lot
of difference (beating < 1hz).   for finer resolution (still > range of human
 hearing)
dividing the 16-bit number by 4 gives you from 0 to 16384 hz with the maximum
 you'd be off from
the real pitch would be 1/4 hz. (if i'm not wrong, 14080hz is 5 octaves above
 the A above
middle C -- plenty high?)
 
        Note-on velocity notenumber             - old way
        Note-on velocity lo-byte hi-byte        - new way
 
would just cost one extra byte.  you wouldn't need pitch bending.
 
that would be great for monophonic instruments -- just play the pitch specified.
 
for polyphonic instruments, you'd have to say which note was bent to the current
 pitch.
 
as long as we're dreaming, why not specify volumes real-time (assuming a
 controller
could be made that allowed both pitch and volume to be specified).  say
you've got 8 total notes possible.  then every pitch message boils down to
 
        note-number volume lo-byte hi-byte
 
if a note's volume ever reaches zero, the next new note is assigned to it.
when each new note is assigned, envelopes are triggered (as in note-on) but
are modulateable by the real-time volume byte in the pitch message. obviously
pitch is specified real-time also.  all for only 4 bytes!  yeah! that's the
 ticket!
 
sorry, i've got to go.  Cindy Crawford is waiting in my Lamborghini to take
me to my Lear jet.  .   .     .      .      .       .        .
 
 
kevan
=========================================================================
Date:         Tue, 25 Oct 1994 18:24:45 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Joe McMahon <xrjdm@FARSIDE.GSFC.NASA.GOV>
Subject:      How to get the digest
 
>I'd rather subscribe to the digest (I can't keep up with all the postings).
>Thanks
 
I will do it for you. For anyone else who is interested; to receive the
day's messages as one big bundle sent out at midnight listserv-time, send a
 
 
SET EMUSIC-L DIGEST
 
to listserv@auvm.american.edu.
 
 --- Joe M.
 
--
"Don't be absurd. We're scaring the locals with music for fun, not profit."
(Lord Cthulhu, in Savage Henry #1)
=========================================================================
Date:         Tue, 25 Oct 1994 18:24:48 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Joe McMahon <xrjdm@FARSIDE.GSFC.NASA.GOV>
Subject:      Re: Ethernet???
 
>          Can someone help me out here, I am fairly new the EMusic-L
>          mailing list, and I don'st understand what this Ethernet
>          thing is.  Is this a network that you can sign on with your
>          computer that broadcasts MIDI Signals????  I'm confused!
 
Not exactly. Ethernet per se is just a higher-speed networking protocol
which has drawbacks of its own. No one has implemented it in a commecial
product due to these drawbacks. As a vehicle for communicating between
computers, though, it is quite good. The discussion is indeed probably
wandering awfully far afield from the actual topics of this list.
 
 --- Joe M.
 
--
"Don't be absurd. We're scaring the locals with music for fun, not profit."
(Lord Cthulhu, in Savage Henry #1)
=========================================================================
Date:         Tue, 25 Oct 1994 18:24:52 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Joe McMahon <xrjdm@FARSIDE.GSFC.NASA.GOV>
Subject:      MIDI limitations do not equal synth limitations
 
The thing I see as a point which may be obvious to everyone by now, but
what the heck:
 
The MIDI implementation has it own problems. However, the principle
bottleneck in MIDI networks is not usually the data transfer, but the data
response. Consider:
 
1) Many synths do *not* simply ignore data which is meaningless to them. In an
   overload condition, in which many, many improperly formed messages are being
   sent, many synths crash or hang, rather than just ignoring the overruns.
   The fact that this can happen at all points to the bottleneck being in the
   processing and not the propagation.
2) Let's make an estimate, which is probably not accurate, but should be in the
   ballpark: With an ability to deliver about 31 thousand bits per second,
   a maxed-out MIDI network should be able to deliver around thirty events per
   second on 16 channels (and this is assuming 1 wire, 8-byte messages). How
   many synths can process 30 events per second in close to real-time?
 
The classic question when tuning the performance of any computer network
is, "Fix the worst response time first." In the case of MIDI, that's almost
always going to be how fast the synthesizer processes events, rather than
how fast events can flow.
 
The big weakness right now of MIDI *is* it's handling of continuous
controllers; though perhaps (much) faster processors might be able to
interpolate a curve in real-time to respond to controller motion and send
that as a sysex message...
 
 --- Joe M.
 
--
"Don't be absurd. We're scaring the locals with music for fun, not profit."
(Lord Cthulhu, in Savage Henry #1)
=========================================================================
Date:         Wed, 26 Oct 1994 10:43:54 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         David Rodger <MUSDR@LURE.LATROBE.EDU.AU>
Subject:      MIDI volume
 
Hi MIDIphiles and MIDIphobes,
On Oct 25, David O'Donoghoe wrote:
>>  allowing a long note to change volume in mid note
 
Controller 7 already does this (doesnt it?)
 
Controller 7 will change the volume of all notes on that particular channel.
Only if one note is playing will a _single_ note be changed.
 
Regards, David Rodger <musdr@lure.latrobe.edu.au>
=========================================================================
Date:         Wed, 26 Oct 1994 09:28:35 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Joe McMahon <xrjdm@FARSIDE.GSFC.NASA.GOV>
Subject:      It's OK - you don't have to unsub
 
>Hi I was ask to "unsubscribe" and towards the bottom of the message it
>mentioned something about "multiple recipents". As far as I know, I only ask
>to subscribe once. I enjoyed the mail I received, however, as I am new to the
>internet and its ediquitte, I am unsubscribing, as I have been instructed to
>do. But I really wish I didn't have to. Bye
 
No, really, you don't have to! If you want me to put you back on, just let
me know.
 
*** PEOPLE: PLEASE write to me (xrjdm@farside.gsfc.nasa.gov) or Metlay
(atomic@netcom.com) if you're having trouble. I don't want our new users to
be confused like this!
 
 --- Joe M.
 
--
"Don't be absurd. We're scaring the locals with music for fun, not profit."
(Lord Cthulhu, in Savage Henry #1)
=========================================================================
Date:         Wed, 26 Oct 1994 09:00:19 CST
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Harry F.P. Haecker" <haeckerh@OSPREY.NWRC.GOV>
Subject:      Re: MIDI limitations do not equal synth limitations
 
Joe M. sez:
 
>The big weakness right now of MIDI *is* it's handling of continuous
>controllers; though perhaps (much) faster processors might be able to
>interpolate a curve in real-time to respond to controller motion and send
>that as a sysex message...
 
Will ZIPI have more increments, a la pitch bend with its 8192 steps, for
continuous controllers?  I've never liked the fact that I can hear the
individual points between 0-127 on, say, a filter opening (esp. noticeable
when the filter is scanning a wide range of frequencies).  That's one
reason that I like analogue synths (one of a total of only 4-5?).
 
HFPH
=========================================================================
Date:         Wed, 26 Oct 1994 15:24:59 +0000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Pall Haraldsson <pallha@RHI.HI.IS>
Subject:      Re: Simple mic question.
In-Reply-To:  <199410211833.SAA07416@eldborg.rhi.hi.is> from "Jacek Latanowicz"
              at Oct 21, 94 06:19:19 pm
 
> >        The formula is as follows:
> >        Freq response required = 0.5*f(s) + a little bit more.
> >where f(s) is your sample rate.
> >
>    Hi.
>
>   I think You mixed two things together. The 'response frequency' and
> 'samplig frequency'. The response frequency should be more or less but
> aquedate to ear perception 18-18000Hz. And the sampling frequency should
> be at least twice as great as the highest sampled frequency. So :
> 18000 * 2 = min. 36000 Hz. It was defined by Nequist and if YOu will drop
> below that the sound reproduction might be not clear.
>
> The exact factor is more than 2. I think it's 2.3 or 2.27 but I'm not sure
> right now.
>
> Jacek L.
>
> Ps. If You want further details I can explain it from the acoustical
> point of view - why should it be 2.3 times higher.
 
From what I've heard it's exactly 2 but that is just theoretically isn't it?
I've heard that sampling rate has to be a little higher because of imperfect
filters, i.e. if you would have a perfect filter the factor would be 2. Is
that right? And, if it is then doesn't the sampling rate have to get higher
with worse filters? And is it realy the factor that changes, i.e. is it
 
sample_rate = (2 + delta) * max_freq.
or
sample_rate = 2 * max_freq. + delta
 
and how do you calculate the delta?
 
And, another thing I've always wanted to ask but wasn't sure who to ask,
how does a (digital) mixer mix digital samples? Let's say you've got two
16 bit samples which you want to mix together (with equal volume) wouldn't
you get one 16 bit sample but lose 1 bit from each sample, i.e. the 16 bit
one is two 15 ones combined? => And if it where 4 16 bit ones you would
get 4 15 bit samples combined in one 16 bit? Etc, etc...
 
Pall Haraldsson (pallha@rhi.hi.is)
=========================================================================
Date:         Wed, 26 Oct 1994 11:48:16 EDT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Eric Harnden (Ronin)" <harnden@DENALI.PHYSICS.AMERICAN.EDU>
Subject:      midi pitch resolution
 
someone suggested using 2 midi bytes for note control.
nope.
first, please don't forget that only seven bits of any data byte
are usable (the msb is a flag), so the resolution of two data bytes
is fourteen bits, for a maximum value of 16384. 16000 is not an
unreasonable top note (i think that the top midi note is g8, which
ought to be in that area), but if each midi note is to represent an
integer frequency, then not much is left for smooth pitch modulation.
a concert A, for instance, is 440Hz. A# is supposed to be 466.16Hz,
disregarding mean/just intonation arguments. if you use only integers,
then that's 466Hz, which is about 27 cents off. pretty bad tuning. the
only thing to do then is to limit the top and bottom range of expressed
values, reserving the extra data to improve resolution between notes. but
with only fourteen bits you really don't have enough to fill in all the
gaps. so go to three bytes?
naah. the hierarchical tuning structure of midi as it is begins to make a
little sense. one byte for note number (arbitrary pitch, mind you... most
new synths and a couple of old ones allow you to set your own actual
note number-to-pitch map), two bytes (when fully implemented) of pitch
bend between each. not bad, really.
 
--------------------------------------------------------------------------------
Return-Path: <>
Date: Wed, 26 Oct 1994 10:37:43 EDT
 
 
----- End forwarded message
=========================================================================
Date:         Wed, 26 Oct 1994 12:02:16 EDT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Eric Harnden (Ronin)" <harnden@DENALI.PHYSICS.AMERICAN.EDU>
Subject:      syntha brevis
 
well, actually, i am one of those people who tweaks forever, but i don't
really have a problem with my music moving from one setup to another.
i have a great deal of faith in serendipity. if it sounds different, i'll
probably like it. that is, after all, one of the main reasons i took up
electronic instruments to begin with. they sound different. and i'm
more and more appreciative of the ephemeral nature of a given sound.
i like to think of it as jazz. what is happening now is just what is
happening now. later, it'll be something else. of course, that's also
why i bought a dat recorder. sometimes i want to be able to make a record
of just what's happening now.  but that's still within the spirit of the
thing, i think. only a very few composers are so hardline about the
musical meaning of the moment that that they continue to refuse to be
recorded. and in classical music that ephermerality is long since gone.
we of course like to think it's there, and so collect different performances
of the brandenburgs, for instance. but be aware that what you get on cd
these days was probably never actually performed as you hear it. it's
a digital edit of many, many takes. the sound you hear is, in a sense,
not just ephemeral, but not even real. so... synths come, synths go.
that's ok. sometimes one comes along that i really want to work with for
a while. and my music then evolves. i've stopped using certain pieces
of equipment entirely. my prophet 5 isn't going to last another ten years.
i'll really miss the jx8p if it blows any time soon. but maybe by then i'll
be able to afford a waldorf... and so on.
 
<--- Entropy Always Wins, And I Like To Be On The Winning Side --->
  Eric Harnden (Ronin) harnden@physics.american.edu (202)885-2746
=========================================================================
Date:         Wed, 26 Oct 1994 12:48:32 LCL
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "Colin T. Smith" <Colin_T._Smith@BOPS.VOA.GOV>
Subject:      Re: Nyquist
In-Reply-To:  TCPG1:BOPS:VOA's message <26-Oct-94 11:27:54 EDT>
 
>From what I've heard it's exactly 2 but that is just theoretically isn't it?
 
The sampling rate according to the Nyquist theory is approximately 2.2 times the
 highest sampled frequency.
 
Think about this, and this is not the real reason for the 2.2 times sampling
 rate. Take a 20kHz square wave and sample it at exactly  40,000 times a second.
  There is a possiibility that if the sampling clock falls exactly when the
 sampled wave crosses the zero point, the resulting sample will be a flat line
 (no audio)  Now this is extremely unlikely (1 in 40,000) but is theorically
 possible. Anything over  a 2 times sampling rate would allow for the squre wave
 to be accurately sampled.
 
>I've heard that sampling rate has to be a little higher because of imperfect
>filters, i.e. if you would have a perfect filter the factor would be 2. Is
>that right? And, if it is then doesn't the sampling rate have to get higher
>with worse filters? And is it realy the factor that changes, i.e. is it
 
Yes, not so much imperfect filters but the filters are analog and it is
 impossible to get an analog filter to attenuate instantly to -infinity.
 
Quoting from ºPrinciples of Digital Audioº by Ken C. Pohlman ºto satisfy the
 sampling theorem, manufacturers must design a low pass filter into any
 digitizing system, placing it first in the signal chain.  Because these analog
 filters cannot cut off the frequency precisely at the frquency the theorem
 demands, a guard band is employed.  The filter¹s cutoff frequency is begun at a
 lower frequency, allowing for the filter to sufficiently attenuate the signal.º
 
To reproduce a full 20kHz frequency and allow room for the filter to fully
 attenuate the signal, it is sampled by 2.2 times the frequency desired so the
 cutoff freq can begin at 20kHz and not before it.  if you multiply 20kHz by
 2.205 you come up with 44.1kHz, heh look ma, CD¹s!  I dont know if this
 standard is a result of the filters used at the time all of this digital stuff
 started and the filters have gotten better,  but because a standard was
 established we have to live with what is out there,  or if its the theorical
 limit of analog filters or......
=========================================================================
Date:         Wed, 26 Oct 1994 13:17:56 EDT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "william.b.fox" <wbf@ALOFT.ATT.COM>
Subject:      Re: Simple mic question.
 
From previous posts:
> >From what I've heard it's exactly 2 but that is just theoretically isn't it?
> I've heard that sampling rate has to be a little higher because of imperfect
> filters, i.e. if you would have a perfect filter the factor would be 2. Is
 
The (Harry) Nyquist Theorem states that you must sample at two times the
highest frequency of interest to capture the waveform.  For audio that
tops out at 20kHz, that means the sample frequency must be 40kHz to
faithfully capture a 20kHz sinewave.  Lower the sample frequency causes
aliasing.  This assumes a perfect LPF which has unity gain from 0Hz to
20kHz and a gain of zero above 20kHz.  This is why the CD sample
frequency is 44.1kHz.  Filters are not ideal.  By raising the sample
frequency (the orignal meaning given to oversampling years before the
CD!), we get a guard band between the audio and sampling noise, thus
relaxing the LPF anti-aliasing filter requirements.
 
Pall Haraldsson asks:
> And, another thing I've always wanted to ask but wasn't sure who to ask,
> how does a (digital) mixer mix digital samples? Let's say you've got two
> 16 bit samples which you want to mix together (with equal volume) wouldn't
> you get one 16 bit sample but lose 1 bit from each sample, i.e. the 16 bit
> one is two 15 ones combined? => And if it where 4 16 bit ones you would
> get 4 15 bit samples combined in one 16 bit? Etc, etc...
 
Hmm...  If two samples being added both have a value of 0xFF, then the sum
is 0x1FE.  Two 16 bit words require a 17 bit bit register to prevent
overflow or data corruption.  That is why a digital mixer and your
favorite FX box use 18 or 24 or larger registers for internal
operations.
 
Bill Fox        wbf@aloft.att.com
 
NEW GRAND UNIFIED THEORY DISCLAIMER: The Manufacturer may
technically be entitled to claim that this product is ten-dimensional.
However, the Consumer is reminded that this confers no legal rights
above and beyond those applicable to three-dimensional objects, since
the seven new dimensions are "rolled up" into such a small "area"
that they cannot be detected.
=========================================================================
Date:         Wed, 26 Oct 1994 22:27:48 +0100
Reply-To:     alan@cycle.demon.co.uk
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Alan Fitch <alan@CYCLE.DEMON.CO.UK>
Organization: Home
Subject:      Re: Nyquist
 
In article  <9410261704.aa28845@post.demon.co.uk> Colin_T._Smith@bops.voa.gov
 (you) wrote:
 
> >From what I've heard it's exactly 2 but that is just theoretically isn't it?
>
> The sampling rate according to the Nyquist theory is approximately 2.2 times
 the
>  highest sampled frequency.
>
 
I think everyone's getting confused here - Nyquist requires a factor of 2
 (exactly),
not 2.2.
 
> Think about this, and this is not the real reason for the 2.2 times sampling
>  rate. Take a 20kHz square wave and sample it at exactly  40,000 times a
 second.
 
This isn't quite right as a 20kHz square wave has components above the Nyquist
frequency - the Nyquist theorem simply says that a signal with frequency
components up to fs/2 can be reconstructed perfectly with a sample rate >=
fs.
 
A 20kHz square wave has components to infinity (or at least a long way...);
for instance the 1st partial of the squarewave would be only 9dB down
at 60kHz.
 
If you want to sample a signal at exactly fs/2 it _must_ be a pure sinewave.
 
> >I've heard that sampling rate has to be a little higher because of imperfect
> >filters, i.e. if you would have a perfect filter the factor would be 2. Is
> >that right? And, if it is then doesn't the sampling rate have to get higher
> >with worse filters? And is it realy the factor that changes, i.e. is it
>
> Yes, not so much imperfect filters but the filters are analog and it is
>  impossible to get an analog filter to attenuate instantly to -infinity.
 
Digital filters are also imperfect.  The reason is that it is not possible to
construct an infinitely fast roll-off filter.  A "brick wall" in the
frequency domain corresponds to an infinitely long impulse response in the
time domain.  For an analogue filter, the impulse response would eventually
fall below thermal noise, and for a digital filter the response would fall
below the precision of the calculations in the filter.
 
>
> Quoting from Principles of Digital Audio by Ken C. Pohlman to satisfy the
>  sampling theorem, manufacturers must design a low pass filter into any
>  digitizing system, placing it first in the signal chain.  Because these
 analog
>  filters cannot cut off the frequency precisely at the frquency the theorem
>  demands, a guard band is employed.  The filters cutoff frequency is begun at
 a
>  lower frequency, allowing for the filter to sufficiently attenuate the
 signal.
>
 
True.
 
> To reproduce a full 20kHz frequency and allow room for the filter to fully
>  attenuate the signal, it is sampled by 2.2 times the frequency desired so the
>  cutoff freq can begin at 20kHz and not before it.  if you multiply 20kHz by
>  2.205 you come up with 44.1kHz, heh look ma, CDs!
 
Complete coincidence, I'm afraid!  The reason for the 44.1kHz rate dates from
 early
digital recorders based on TV technology.  44.1kHz was a multiple of a TV
rate in NTSC I believe.
 
The main design constraint in making filters very rapid cutoff is that it
 introduces
severe phase distortion at the point of cutoff.  Hence a compromise has to be
reached between achieving sufficient attenuation at fs/2, without throwing away
half the signal (due to a slow cut-off implying a low corner frequency), or
introducing loads of ringing and phase distortion.  Of course most modern
DACs and ADCs are oversampled for precisely these reasons, i.e. by oversampling
to 88kHz, the filter design problem is drastically simplified.
 
I hope this helps
                  regards,
                            Alan
 
 
--
Alan Fitch
Tel.:           +44 (0)1703 472342 (voice)
E-mail:         alan@cycle.demon.co.uk
 
Thank god it's not Christmas.
=========================================================================
Date:         Wed, 26 Oct 1994 21:22:10 -0400
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Wesley Felteau <wfelteau@ST6000.SCT.EDU>
Subject:      Re: Mac Av digital audio and disc space
In-Reply-To:  <9410211533.AA08633@st6000.sct.edu> from "Stefan Scheffler" at
              Oct 21, 94 03:50:55 pm
 
The only way that I know how to get more than 2 tracks of dig snd on an AV
Mac is to use a rack mount product by Digidesign called Session 8.
This DSP allows one to record up to 8 tracks digital and mix it down
to two.
The bottleneck for recording more than 2 tracks lies in the Mac architecture
(and PC arch) therefor one must use an external DSP.
 
Wes  N/S Productions Atlanta GA
=========================================================================
Date:         Thu, 27 Oct 1994 13:50:59 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         David Rodger <MUSDR@LURE.LATROBE.EDU.AU>
Subject:      Re: Nyquist
 
On 26 October, Alan Fitch wrote:
 
> Complete coincidence, I'm afraid!  The reason for the 44.1kHz rate dates
> From early digital recorders based on TV technology.  44.1kHz was a multiple
of a TV rate in NTSC, I believe.
 
This may be true, but is no longer so.  The nominal NTSC rate in the early
(black and white) days of U.S. TV was 30 frames per second.  If we wanted
sample accurate synch with video then 44.1kHz would be fine.  That's 1470
samples per frame.  But when colour came in, there was some obscure (meaning,
I don't know what it was!) technical reason that exactly 30 frames per
second couldn't be used.  So they switched to 29.97 frames per second; i.e.,
0.1% slower.
 
Problem!  All the old shows were at 30fps and if shown at 29.97fps would
run 3 seconds and some frames long for every hour (can't remember the exact
figure).  So some had to be cut out.  A standard way of resolving the old
frame rate ws found and results in "drop-frame" rates.  While no actual
video frames are dropped (or deleted in any way), some of their numbers
are, so that the frame count at the end of an hour will resemble the
real time of the hour.  For example, if a show begins at 00:00:00:00 and
runs at 29.97fps, it is still shot shot at 30frames per "second".  To keep
the count more or less in synch with actual running time, every so often
a frame number (not an actual frame) is dropped.  The result is that the
show will end on frame count 01:00:00:00 and the actual running time
will be exactly 1 hour.  This is referred to as 29.97 drop-frame.
 
Complicated, isn't it?  There's a terrific article explaining the ins
and outs of the various time code frame rates, in their "drop" and "non-
drop" (or "full-frame") versions and the ways of cross-resolving time
code between different media, in a special edition of Mix magazine (U.S.)
sometime during 1993 (May, I think).  It's by Larry Blake and is very
good.  I think he's written a book about time code since, which might
be available on the MIX Bookshelf.
 
So, how does this relate to sampling?  Well, with dropframe time code,
you no longer have a whole number of samples per frame and the audio
will actually be going out of synch with the video until a drop in the
frame count occurs (I don't recall how often).  Sometimes, lower or
higher sampling rates (44.056, 44.144, 47.952 kHz) are used so that the
various media (video, audio, etc.) can be resolved during post-production.
 
It's really much easier here in Australia, and in Europe, where the
standard time code is 25 frames per second for film and video.  So we
always have a whole number of audio samples per frame and always stay
in synch.  Of course, with so much material being sold across the world,
we have to deal with U.S. programs with dop and non-drop 29.97 and 30fps
and people in the U.S. have to deal with those programs using lower
frame rates too.  Unfortunately, there's no simple way around this, at
least until we all switch exclusively to some agreed standard for
disk-based non-linear audio and video systems!  (Not likely!)
 
Anyway, if you're only worried about audio, all this doesn't matter.
In fact, 48kHz is a more common audio-only sampling rate (as used by
reel-to-reel and cassette-based digital machines), although some
machines will let you select.  There's sometimes talk of really high
sampling rates (e.g. 96kHz) so that the shape of high-frequency
waveforms is preserved better.  Can we hear the difference?  I don't
know.  I'm just happy if the damn thing sounds good!
 
Happy sampling and video editing,
David.
=========================================================================
Date:         Thu, 27 Oct 1994 01:42:31 -0400
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         john rossi <jrossi@BGNET.BGSU.EDU>
Subject:      Re: Mac Av digital audio and disc space
In-Reply-To:  <199410270135.VAA16763@falcon.bgsu.edu>
 
On Wed, 26 Oct 1994, Wesley Felteau wrote:
 
> The bottleneck for recording more than 2 tracks lies in the Mac architecture
> (and PC arch) therefor one must use an external DSP.
>
 
I don't know enough about Macs to comment, however, I fail to see how the
PC arcitecture precludes recording more than two tracks.  Of course you
would need A/Ds and the corresponding sample and holds but I see nothing
which limits the number of channels you can record except, possibly, bus
speed.  With Local Bus speeds of up to 50MHz, and PCI speeds at 33 MHz,
you would be able to record a hell of a lot of channels if you, say, ran
under dos and weren't generating spurious interrupts.  I mean, ADATs and
DA-88s get 8 comfortable channels of 48K sampled data into a littl more
than 4MHz bandwidth.  Of course a computer has a lot of overhead
considering disk drive access, but even if it took as many as 20 machine
cycles to sample one channel, you still have more than twice the
bandwidth of an ADAT,l even in the worst case.  Even if you assume that
the disk drives would be the rate limiting step, 8 channels at 48KHz
would only require sustained data rates of about 3.8MHz, hardly taxing
for a decent SCSI drive.
 
John 3
=========================================================================
Date:         Thu, 27 Oct 1994 03:19:11 EDT
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Phillip Djwa <djwa@IEAR.ARTS.RPI.EDU>
Subject:      Re: Mac Av digital audio and disc space
In-Reply-To:  <9410270143.AA04208@iear.arts.rpi.edu>; from "Wesley Felteau" at
              Oct 26, 94 9:22 pm
 
Geez,
I'm not sure whether I missed the first posting re:digital audio on AV
macs, but I have a 660AV and I have 6-trks digital audio with DECK II
from OSC in San francisco(or is it San potrero??) My friend with an 840AV
has 8trks with Deck II. It is a sometimes flaky program, but nothing
as flaky as protools, so I'm happy.
 
phillip djwa
djwa@iear.arts.rpi.edu
=========================================================================
Date:         Thu, 27 Oct 1994 03:54:28 -0500
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Arne Claassen ISE <claassen@EBS330.EB.UAH.EDU>
Subject:      Re: Mac Av digital audio and disc space
In-Reply-To:  <9410270543.AA10491@ebs330> from "john rossi" at Oct 27,
              94 01:42:31 am
 
>
> On Wed, 26 Oct 1994, Wesley Felteau wrote:
>
> > The bottleneck for recording more than 2 tracks lies in the Mac architecture
> > (and PC arch) therefor one must use an external DSP.
> >
>
> I don't know enough about Macs to comment, however, I fail to see how the
> PC arcitecture precludes recording more than two tracks.  Of course you
> would need A/Ds and the corresponding sample and holds but I see nothing
> which limits the number of channels you can record except, possibly, bus
 
I have to agree. The fact that there are various programs that let you do
multi-track work on AV machines shows that it can deal with multiple tracks
in real time. The real problem is that there is only that one stereo pair
for input. Same problem with an audiomedia II. You can record only as many
tracks as there are inputs. A problem that the Session 8 interface does
take care of, but not because of more DSP power, but more A/D converters.
 
--
Arne F. Claassen           <claassen@ebs330.eb.uah.edu>  <psycho@io.uah.edu>
On the Web:           <http://isl-garnet.uah.edu/psychohttp/psychohome.html>
-----------------------------------------------------------------------------
"[I]f a license was required to drive your synth, there might be less of a
need for these newbie-type questions - then again, we'd have to deal with
a Department of Transportation (Workstation?) - aaahhhhh!" - Jeff Schultz
=========================================================================
Date:         Thu, 27 Oct 1994 22:22:00 +1300
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         John Thomsen <THOMSENJ@SCHOOLS.MINEDU.GOVT.NZ>
Subject:      Assistance required with Mellotron MUART/Spirit
 
To the Internet, from the Antipodes, greetings.
 
In other words, G'day from Down Under!
 
I have a problem.
 
I purchased from Radio Shack in winter '91 a Mellotron MUART MIDI
controller card/box and Mellotron Spirit sequencer software.  I had
had no problems with it, until 1630 Fri 21 Oct 94 NZDT.  All of a
sudden, the sequencer software would no longer work.  (A big problem,
because I was in the middle of a recording session.)
 
On Tue 25 Oct 94 I attempted to call the Mellotron Technical Support
Group at (813) 572 6356.  I let it ring probably 30 times, then gave
up.  (The New York time was about 1530 Mon 24 Oct 94.)  Then I tried
the number for Mellotron Digital Corporation.  I got the following
recorded message:
 
   'Beep-beep-beep.  The number you have dialed, 944 6789, in area
   code 516, has been disconnected.  No further information is
   available.  The number you have dialed, ...'
 
So, my questions for the Internet are manifold:
 
   1.  Does anybody know what happened to Mellotron?  NY Directory
       Assistance doesn't have any listing for any Mellotron.
   2.  If they're still around but have changed their name/location/
       etc., does anybody know how I can get in touch with them?
   3.  If they don't exist anymore, does anybody have a copy of the
       Mellotron Spirit software that they could please zip or arj and
       uuencode and mail to me?  I've traced the problem to the
       software, but both my original and backup disks have been
       toasted, and I'm really up the proverbial estuary without means
       of locomotion.  I *desperately* need my software up and
       running.
 
You're probably thinking why I just don't get a SoundBlaster with MIDI
capability, the right hardware to hook it up to my synth, and another
music program.  The reasons:
 
   1.  Spirit uses a nonstandard file format.  The MIDI information is
       still the same, but the format's weird.
   2.  I don't know how a proper MIDI file works, so I can't write a
       conversion program.
   3.  I can't afford at the moment to get another set of hardware.
   4.  I can't afford more software.
   5.  Besides, I don't think that I could find anything that would
       work on my ancient 8-MHz 8086 with 640K RAM and two floppies
       (360K and 720K).
 
Further: I apologise to all those out there who subscribe to more than
one mailing list who have seen this twice or more.  I'm so desperate
that I've posted this anywhere I think it might do some good.  Does
anybody know of a listserv for MIDI musicians I could try posting to?
 
The canonical list of lists that this went to is as follows:
   WIN3-L
   ACTMUS-L
   ALLMUSIC
   AT-PROF
   AUDIO-L
   BRASS-L
   BLUES-L
   CLASSM-L
   EARLYM-L
   EMUSIC-L
   ETHMUS-L
   GEACMUS
 
*PLEASE* don't flame me, as I don't know the address for a MIDI music
list, and I figure that anybody out there could have this software.
Just sending this to one listserv wouldn't reach as wide an audience
as sending to multiple lists. Usenetters can be thankful for the fact
that I don't have access to Usenet (but this will change in a month or
so... watch out! ;-)
 
Also: Please reply personally to me.
 
Hoping sincerely that someone out there can help me,
 
John Thomsen
Musician and chief programmer at Weresoft NZ
email address: thomsenj@schools.minedu.govt.nz
=========================================================================
Date:         Thu, 27 Oct 1994 20:53:51 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Daniel Murphy <dmurphy@PIONEER.AS.EDU.AU>
Subject:      Hi! What's This List Like?
 
Hi all!
 
I just subscribed and was wondering who was on the list and what you're all
like. I suppose you get a heap of these type of messages so I'll tell you a
bit about myself.
 
I'm from Australia. I play the pipe organ and piano. I LURVE movies. I live
on the beach. I like springboard diving, skiing (both types) and swimming.
My favourite artists - at the momenet - are Madonna, Prince and
Soundgarden. That's about it I suppose.
 
Now it's your turn!
 
 
 
Daniel.
 
 
  ==================================================
  |             Daniel Murphy                      |
  |              3 Illawong Street                 |
  |               Point Cartwright                 |
  |                Buddina Beach                   |
  |                 Queensland 4575                |
  |                  AUSTRALIA                     |
  |                                                |
  |           dmurphy@pioneer.as.edu.au            |
  ==================================================
=========================================================================
Date:         Thu, 27 Oct 1994 12:04:04 GMT0BST
Reply-To:     M.Rootes@shu.ac.uk
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Martin Rootes <M.J.Rootes@SHU.AC.UK>
Organization: Sheffield Hallam University
Subject:      Re: Nyquist
 
> From:           David Rodger <MUSDR@LURE.LATROBE.EDU.AU>
> ......
> It's really much easier here in Australia, and in Europe, where the
> standard time code is 25 frames per second for film and video.
I'm not sure this is correct, I have always been told that the frame rate for
film is (universally) 24 frames per second. The frame rate for televison was
based on the mains AC frequency hence Europe & Australia 50Hz leads to 25 fps,
America 60Hz leads to 30 fps (originally). I've been told that the film frame
rate of 24 was decided as it can be divided by multiples of 2 and 3, however
this could just be an urban myth.
 
  Martin.
 
------------------------------------------------------------------------------
Martin Rootes - Senior Systems Programmer/Analyst, Sheffield Hallam University
Email :         M.Rootes@shu.ac.uk
------------------------------------------------------------------------------
  As this is not a real signature, the opinions above may not even be mine!
------------------------------------------------------------------------------
=========================================================================
Date:         Thu, 27 Oct 1994 12:03:58 -0400
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Sonneck Society for American Music <monty@ACCESS.DIGEX.NET>
 Subject: American Music Network
Message-ID: <Pine.SUN.3.91.941027115826.16270B-100000@access4.digex.net>
MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII
 
                                        October 27 1994
 
SONNECK SOCIETY FOR AMERICAN MUSIC ANNOUNCES THE AMERICAN MUSIC NETWORK
 
     The Sonneck Society for American Music announces the establishing of
the American Music Network on the Internet.  The Network can be accessed
by the following commands:
          gopher gopher.tmn.com.
          select #5 Arts Wire
          select #6 American Music Network.
     The Network has been established as a service to the public and
interested organizations to promote the performance and study of
American Music.
     There are currently 2 databases available on-line.  These are:
          Coming Events in American Music
          American Music Week Events
     Several other databases are being developed for access by the
public.  These are:
          Speaker's/Performer's Bureau for American Music
          Important Dates in American Music History
     Members of the Sonneck Society may submit data to be added to any
of the databases.  Members of the public may search and read any of the files
using gopher or other Internet tools.  A dial-up line is planned to be
installed for public access to the American Music Network in the future.  It
is also planned to establish a mailing list (listserv) for American Music
in the future.
     Any organizations sponsoring events as part of American Music Week,
normally the first week in November (November 6-13 1994), may submit their
events directly to the Network. Information on format for submittals
and membership is available on-line at gopher gopher.tmn.com,
as described above.
     It is hoped that this Network will become a clearinghouse for all
areas of American Music.  Performers will be able to offer their talents
to the public, worldwide.  There is no charge for the service, other than
membership in the Sonneck Society ($50/year).  Promoters and program
managers will be able to find performers or events of interest on
American Music.  The public and media will be able to find events on
American Music easily and conveniently.
     Questions on the American Music Network may be sent to:
          Bill Everett, Chair, Public Relations Committee
          Sonneck Society for American Music
          email: <zzever@acc.wvacc.edu>
=========================================================================
Date:         Thu, 27 Oct 1994 12:42:49 -0400
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Sonneck Society for American Music <monty@ACCESS.DIGEX.NET>
Subject: Correction of Announcement on American Music Network
Message-ID: <Pine.SUN.3.91.941027122947.17719B-100000@access4.digex.net>
MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII
 
Please correct email address given in previous announcement.
 
     Questions on the American Music Network may be sent to:
          Bill Everett, Chair, Public Relations Committee
          Sonneck Society for American Music
          email: <zzever@acc.wuacc.edu>
=========================================================================
Date:         Thu, 27 Oct 1994 18:18:27 +0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Stefan Scheffler <stefan@MMS-GMBH.DE>
Subject:      Re: Mac Av digital audio and disc space
 
>under dos and weren't generating spurious interrupts.  I mean, ADATs and
>DA-88s get 8 comfortable channels of 48K sampled data into a littl more
>than 4MHz bandwidth.  Of course a computer has a lot of overhead
>considering disk drive access, but even if it took as many as 20 machine
>cycles to sample one channel, you still have more than twice the
>bandwidth of an ADAT,l even in the worst case.  Even if you assume that
>the disk drives would be the rate limiting step, 8 channels at 48KHz
>would only require sustained data rates of about 3.8MHz, hardly taxing
>for a decent SCSI drive.
All is true what you wrote, but you did not look at the problems at large.
You made the common mistake to forget that the general purpose computer has
to do other tasks as well.
 
The CPU, The Heat
Todays CPU's are fast enough to do tons of sound channels with EQ and other
fancy stuff. Including the production of heat.
 
The bottle neck number one the Operating system
With the current Operating systems the processor, bus and what soever waits
until completing of an I/O request. When the disk can transfer at a rate of
4 Mb/s and the computer reads this amount of data from it, it will take the
computer exactly one second to complete this transfer.
That means your bandwidth is suddently shrinked to 4 MHz and nothing else
had happened in between, no mixing of data, no further transfer to the A/D
converter (though this is done today with DMA sometimes, therefore in
parallel to the CPU) no screen updates, and no user interaction handling.
 
The disks limits
Also there are very few (at least no one I know of) hard disks which can
transfer the data for multiple audio tracks with the above mentioned
transfer rate. Today 1.5 to 2.5 MB/s is more common for 4 track systems (4
* mono or 4 * stereo) and 1.0 to 1.5 MB/s for 8 tracks  (8 * mono). This is
sufficient enough for a standalone hard disk recording system. But if your
general purpose computer with the big two OS's has to do it, it cannot do
anything else.
(Note: The transfer numbers can be increased by using a lot more buffer
memory inside the computer. The disk transfer rate goes down through the
head movement and the disk spinning, by providing more buffer memory bigger
adjacent chunks of data can be read from disk, therefore reducing the head
movement)
 
Why an ADAT/DA-88 can handle it
Systems like the ADAT or the DA-88 get their data as a continous stream
from the tape, therfore the data transfer is constant over the time and
needs less buffering. Also the data processing will not be done by the main
CPU. In fact the "main" CPU in such a system is as powerful as the PC
keyboard controller. Through the dedicated sub systems in such a device it
can perform its task with less overall power and is also more stable.
 
At the mainframe heaters in your todays general purpose computer most heat,
i.e. power, gets lost through inefficient operating system design and
implementation.
 
The Mac OS is currently the most capable OS. It can do more tasks which has
to do with hard disk recording than the other popular system. Also the Mac
AV computer system has an architecture which support this kind of
"multitasking".
(I cannot tell anything about OS/2 it might can get things even better. My
NeXT does, though its not a popular system or OS).
With a proper future OS, even next years OS, this problem can be solved,
but as of today it is not.
 
Be happy to get 2 or four tracks today, tomorrow you will get tons. But
there are different definitions of tomorrow.
 
Ciao,
   Stefan
 
<> stefan@mms-gmbh.de
<> Stefan Scheffler, Software Development
<> Steinberg Soft- & Hardware GmbH
<> Cubase Lite/Score/Audio for Mac/Atari/Windows
=========================================================================
Date:         Fri, 28 Oct 1994 12:53:41 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         David Rodger <MUSDR@LURE.LATROBE.EDU.AU>
Subject:      Re: Hi! What's This List Like?
 
Hello Dan,
welcome to the list from another Australian.  You'll also see David
O'Donoghoe (spelling?), Paul Copeland and Clive McFarlane, among others,
who would seem to be Ozsters, too.  I mention these chaps because they're
very vocal (or should that be literal?).
 
Happy emusic,
David Rodger    <musdr@lure.latrobe.edu.au>
=========================================================================
Date:         Fri, 28 Oct 1994 12:57:53 +1000
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         David Rodger <MUSDR@LURE.LATROBE.EDU.AU>
Subject:      SMPTE and film frame rates
 
On 27 October, Martin wrote:
 
>I'm not sure if this is correct, but I've been told that the frame rate for
>film is (universally) 24 frames per second.
 
I don't know what proportionof filmmakers are using 24 or 25 fps, but, either
way it makes for easier transfers than 29.97 fps or drop frame.
 
As for 24 being chosen for its duple and triple divisibility, I don't know.
Film historians, it's your call...
 
Regards, David.
=========================================================================
Date:         Fri, 28 Oct 1994 10:13:26 +0200
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         "A. Fiorletta" <ccd505@CCD.UTOVRM.IT>
Subject:      Mr. Fourier contest
 
Hello guys!
 
What is the fastest Fast Fourier Transform algorithm that you know?
Please send a mail with your plain C code and you'll enter in the competition.
PLEASE NOTE that you will win nothing but the good feel of helping someone
(me!) in doing its spectrum job faster. You will also win the proud of being
the "Fastest Fourier Transformer" of the list!
 
Many Thanks for your help.
 
Alessandro
=========================================================================
Date:         Fri, 28 Oct 1994 13:02:29 +0100
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Stefan Scheffler <stefan@MMS-GMBH.DE>
Subject:      Re: Mr. Fourier contest
 
>What is the fastest Fast Fourier Transform algorithm that you know?
>Please send a mail with your plain C code and you'll enter in the competition.
>PLEASE NOTE that you will win nothing but the good feel of helping someone
>(me!) in doing its spectrum job faster. You will also win the proud of being
>the "Fastest Fourier Transformer" of the list!
What computer do you need it for. Algorithms which have to speedy has to
take advantage of the CPU architecture. 68K, RISC or DSP based
implementations will not run as fast on a x86 based system and visa versa.
I cannot provide you with ready to go implementations (I get always object
files), but when you know your CPU architecture you will be able to tune
the FFT yourself.
You can also look at the IRCAM or the CCRMA ftp sites, they might have FFT
implementations for various system.
 
You will get "Fastest Fourier Transformer" of the list when you beat a
Motorola 56K DSP with 32 MHz. It needs 3 milliseconds for a 1024 point FFT.
(Maybe its an unfair comparison, the 56K was designed with the FFT in
mind). Anyway, good luck and keep hacking.
 
Ciao,
    Stefan
 
<> stefan@mms-gmbh.de
<> Stefan Scheffler, Software Development
<> Steinberg Soft- & Hardware GmbH
<> Cubase Lite/Score/Audio for Mac/Atari/Windows
=========================================================================
Date:         Fri, 28 Oct 1994 13:53:59 -0700
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Patte Wood <patte@CCRMA.STANFORD.EDU>
Subject:      Russian Electroacoustic Music
 
LECTURE AND CONCERT
OF
RUSSIAN ELECTROACOUSTIC MUSIC
 
at the
Center for Computer Research in Music and Acoustics (CCRMA)
The Knoll, 660 Lomita Drive, Stanford University Campus
 
Andre Smirnov, Director of the Electroacoustic Studio in Moscow, will present a
 lecture regarding his
work on Wednesday, November 2 at 1:15 pm in the Knoll Ballroom.  He will also
 present a short concert
of electroacoustic music in Russia on Thursday, November 3  at 4:00pm in the
 Knoll Ballroom.
 
for information and directions call 415 723-4971.
=========================================================================
Date:         Fri, 28 Oct 1994 14:14:57 -0700
Reply-To:     Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
Sender:       Electronic Music Discussion List <EMUSIC-L@AMERICAN.EDU>
From:         Patte Wood <patte@CCRMA.STANFORD.EDU>
Subject:      Russian Electroacoustic Music
 
If you got this twice, I apologize.
 
 
 
LECTURE AND CONCERT
OF
RUSSIAN ELECTROACOUSTIC MUSIC
 
at the
Center for Computer Research in Music and Acoustics (CCRMA)
The Knoll, 660 Lomita Drive, Stanford University Campus
 
Andre Smirnov, Director of the Electroacoustic Studio in Moscow, will present a
 lecture regarding his
work on Wednesday, November 2 at 1:15 pm in the Knoll Ballroom.  He will also
 present a short concert
of electroacoustic music in Russia on Thursday, November 3  at 4:00pm in the
 Knoll Ballroom.
 
for information and directions call 415 723-4971.
