| hello again
using reson instead of butterbp solved the problem, but i got stuck
to a wierd problem.
if i control say randi with another random signal from linrand inside
timout..reset..rireturn output gets out of range. Commenting timout etc,
makes the output reasonable
????
-matti
mjkoskin@sci.fi
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Date: Mon, 24 Aug 98 15:47:05 BST
From: jpff@maths.bath.ac.uk
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Message written at 23 Aug 1998 19:40:48 -0400
I am getting rather upset by the Leeds Front Page, which seems to have
a number of inaccuracies and is dated. Does anyone know who is
resposible for it? I have not managed to get any correction so far --
even my name is wrong.
==John ffitch
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From: jpff@maths.bath.ac.uk
Subject: Re: more strange thigs
To: nicb@axnet.it, csound@maths.ex.ac.uk
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Message written at 24 Aug 1998 11:18:37 -0400
--- Copy of mail to mjkoskin@sci.fi ---
In-reply-to: <35E1702F.BE55B73B@sci.fi> (message from Matti Koskinen on Mon,
24 Aug 1998 16:52:47 +0300)
References: <35E1702F.BE55B73B@sci.fi>
There was a bug in randi; I have been working over all rand and
filters checking for some bad effects. I think I am nearly done now.
I will check out teh Butterworth filters as well as I thought they
were OK.
==John ffitch
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Date: Mon, 24 Aug 1998 14:15:42 -0400
To: csound@maths.ex.ac.uk
From: Bertolt Sobolik
Subject: Re: PMAX
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Why not just write the values to a file and read them in with GEN 1?
At 10:53 AM -0800 8/24/98, Mark Polishook wrote:
>I have an GEN 2 table with 512 values which I'm reading through using phasor
>and table. Csound only reads through the 1st 150 values of the
>table and then flushes the rest.
>
>Can PMAX be re-set so that I can read the entire table?
>
>!!thanks in advance
>
>-mp
>
>;;;;;;;;;;;;;;;;;;;;;;;;;;;;voss.orc
>
>aseg linseg 0, p3/2, 1, p3/2, 0
>k1 phasor 1/p3
>kvoss table k1 * 512, 2
>asig oscil p4*aseg, p5*kvoss, 1 ; oscillator
>
> outs asig, asig
>endin
>
>;;;;;;;;;;;;;;;;;;;;;;;;;;;;voss.sco
>
>f2 0 512 -2 1.00735501488 1.02188088158 1.01261369844 etc...
>i1 0 48 5000 440
>e
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Date: Mon, 24 Aug 1998 13:09:59 +0000
From: david first
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Subject: Freq/Tempo Strangeness
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Hi -
Can someone explain the following phenomenon?
When I run the orc/sco below at any of the following tempos - 99, 100, 105,
106, 199, 200, 205 (and probably others) the kcps linseg glissandos in
instruments 3 & 4 resolve to their proper (non-beating) pitches. But if I set
the tempo at 98, 101-104, 107, 108, 201, or 207 the pitches _don't_ resolve
properly and varying degrees of beating are heard.
On the other hand, if I take out the linsegs and run the orc/sco with either
set of pitches found in the kcps linsegs then the pitch resolution is fine and
no beating is heard.
Furthermore, there is at least one pair of kcps linseg pitches for which I
have yet to find a "suitable" tempo (618.75 - 594).
What is this hidden (at least to me so far) relationship between
tempo/pitch/linsegs? Could someone please take a minute to check this out?
d1st
Tested w/ both 3.47 & 3.48 beta Mills/PPC
9600/233
OS 8.1
128 MB ram
10 MB allocated to application
-----------------------------------------------------------------------
sr = 44100
kr = 4410
ksmps = 10
nchnls = 2
instr 1
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg p5, p3, p5
aout1 oscil kamp1, kcps, 1
aout2 oscil kamp2, kcps, 1
outs aout1, aout2
endin
instr 2
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg p5, p3, p5
aout1 oscil kamp2, kcps, 1
aout2 oscil kamp1, kcps, 1
outs aout1, aout2
endin
instr 3
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg 275, p3*1/8, 275, p3*1/8, 264, p3*1/8, 264
aout1 oscil kamp2, kcps, 2
aout2 oscil kamp1, kcps, 2
outs aout1, aout2
endin
instr 4
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg 412.5, p3*1/8, 412.5, p3*1/8, 396, p3*1/8, 396
aout1 oscil kamp2, kcps, 3
aout2 oscil kamp1, kcps, 3
outs aout1, aout2
endin
instr 5
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg p5, p3, p5
aout1 oscil kamp1, kcps, 4
aout2 oscil kamp2, kcps, 4
outs aout1, aout2
endin
instr 7
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg p5, p3, p5
aout1 oscil kamp1, kcps, 3
aout2 oscil kamp2, kcps, 3
outs aout1, aout2
endin
-----------------------------------------------------------------------
f1 0 8192 10 1 .5 .333 .25 .20 .1667 0 .125 .1111 .1 0 .0833 0 0 .05333 .05
f2 0 8192 10 1 .5 .333 .25 0 .1667 0 .125 .1111 0 0 .0833 0 0 0 .05
f3 0 8192 10 1 .5 .000 .25 0 .0000 0 .125 0 0 0 .0000 0 0 0 .05
f4 0 8192 10 1 .5 .333 .25 0 .1667 0 .125 0 .1 0 .0833 0 0 0 .05
f5 0 8192 10 1 .5 .000 .25 0 .0000 0 .125 0 .1 0 .0000 0 0 0 .05
f6 0 8192 10 1 .5 .333 .25 0 .1667 0 .125 0 .1 0 .0833 0 0 0 .05
t 0 100
i1 0 32 1000 55 ;1/1R
i2 0 32 1000 110 ;1/1L
i1 0 32 1000 220 ;1/1R
i3 0 32 1000
i1 0 32 1000 330 ;3/2R
i4 0 32 1000
i5 0 32 1000 495 ;9/8R
i7 0 32 1000 742.5 ;27/16R
e
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Date: Mon, 24 Aug 1998 10:27:47 -0700
From: Erik Spjut
Subject: Re: Freq/Tempo Strangeness
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I don't have time to check out your problem but I am 99.99% certain it will
disappear if you set kr=sr. The %@$$*&^$%#^ control frames always end
places other than the exact time you specify UNLESS your duration is an
EXACT multiple of the control frame period. Hence linseg's, expseg's, and
any other useful signal ends at the wrong place. My guess is that your
tempo's give you the exact relationship when things work.
-------------------------------------------------------------------------------
Erik Spjut (pronounce ju as long u or yew) - Associate Professor of Engineering
and Associate Director for Engineering Computing, Center for Design Education
Harvey Mudd College, Claremont, CA 91711-5990 USA
Erik_Spjut@hmc.edu Ph & Voice mail (909) 607-3890 Fax (909) 621-8967
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Date: Mon, 24 Aug 1998 10:53:26 -0800
From: Mark Polishook
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Subject: PMAX
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I have an GEN 2 table with 512 values which I'm reading through using phasor
and table. Csound only reads through the 1st 150 values of the
table and then flushes the rest.
Can PMAX be re-set so that I can read the entire table?
!!thanks in advance
-mp
;;;;;;;;;;;;;;;;;;;;;;;;;;;;voss.orc
aseg linseg 0, p3/2, 1, p3/2, 0
k1 phasor 1/p3
kvoss table k1 * 512, 2
asig oscil p4*aseg, p5*kvoss, 1 ; oscillator
outs asig, asig
endin
;;;;;;;;;;;;;;;;;;;;;;;;;;;;voss.sco
f2 0 512 -2 1.00735501488 1.02188088158 1.01261369844 etc...
i1 0 48 5000 440
e
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Date: Mon, 24 Aug 1998 16:26:39 +0000
From: david first
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Subject: Freq/Tempo Strangeness Solved
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Erik -
Your idea sent me to the right parameters, but, in fact, the opposite - for
whatever reason - turned out to be the answer. At a tempo of 100bpm
the ksmps seems to need to be equal to 100 or greater for there to be what
sounds to me like rock solid beat resolution. Thanks for the relationship
information and the quick reply.
d1st
Erik Spjut wrote:
>
> I don't have time to check out your problem but I am 99.99% certain it will
> disappear if you set kr=sr. The %@$$*&^$%#^ control frames always end
> places other than the exact time you specify UNLESS your duration is an
> EXACT multiple of the control frame period. Hence linseg's, expseg's, and
> any other useful signal ends at the wrong place. My guess is that your
> tempo's give you the exact relationship when things work.
>
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To: csound@maths.ex.ac.uk
From: jose halac
Subject: winsound perf
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hi...
Im trying to use winsound on a gateway 2000, windows based system.
I am using word for orc/sco, and I'm using the toot.orc toot.sco tutorial
files.
On the program window I select the orc (which shows as an icon with a 1/4
note inside, toot, loosing the .orc) and the score file (which shows as a
text icon, toot.sco).
I check remain active and play after calculate.
Then I hit "OK", starts compiling, and after the verbose, it gets stuck on
"sfinit=cannot open", and nothing happens.
What am I missing?
It's my first try on windows based system, so please forgive my primitivism...
thanks a lot
jose h.
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Date: Tue, 25 Aug 1998 02:58:11 +0000
From: Tobiah
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To: Csound Mailing List , toby@rcsreg.com
Subject: Command line alteration of AIFF instrument parameters.
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I have made a command line utility which is capable of manipulating
some of the more obscure parameters found in an AIFF soundfile.
The program is called 'basef' and can be found at:
ftp://198.176.199.138/pub/basef/
This is a Linux ELF binary with source that should compile on others.
I made it because I needed to change the implied base frequency of
aiff format files, but I went on to allow changes to any of the
parameters that reside in the 'instrument' chunk of the file. The
instrument chunk is created if non-existent.
The program is useful to Csound users who use the 'loscil' ugen.
The idea is to record a real instrument at various pitches, into
a bank of AIFF files, updating each one with 'basef' to hold its own
true base frequency. Now when a sample is called from loscil,
Csound reconciles the current sample rate with the AIFF file's
sample rate and base frequency in order to produce the correct
audible pitch specified in the call to loscil!
The program is very new, and not very tested. It works quite well, but
please don't use it on your Master's Thesis without making a copy first.
Oh, and I used a lot of the Csound source code to read and write to the
files.
Toby
-There otta be a law-
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Message-ID: <35E2397F.312DF84B@panix.com>
Date: Tue, 25 Aug 1998 00:11:50 -0400
From: Michael Coble
Reply-To: coble@panix.com
Organization: Smiling Dog Apps
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Subject: Re: IRCAM format
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Ben Jefferys wrote:
>
> Charles Baker wrote:
>
> > Why the heck isn't there *more* emphasis on floating point in
> musical
> > signal processing ? Makes all the difference
>
> Of course it doesn't necessarily have to be floating point. I'd guess
> a large part of the reason why floating point stuff is better is
> because it uses more bits - typically 32/64/80 (in think they're
> the standards). If you used 32 bit ints the quality would improve
> by a similar amount to your standard C floats, although there would
> be subtleties lost in quieter audio compared to using a float due to
> the "logarithmic" nature of floating point. But the 16-24-32 bit
> transition within the integer realm is very significant, and on
> some processors integer performance is superior.
Hello,
Not to belabor the point, but more bits are only a piece of the puzzle. With
16 bit integers, all the sounds need to be interpolated to fit into 65,536
slots. Floating points mean that the samples can be more exact. This would
mean that samples can exist between the slots of the 65,536 points available
in 16 bit samples.
In a lot of computer music this means that noisy crappy sounding stuff can be
generated when creating direct output sounds in 16 bit 44K soundfiles. In
practice, doing the computer generated work in floating point and then
translating that into 16 bit 44K soundfiles makes the same output sound much
more palatable on the ears. Brad Garton used used to recommend that we employ
this method when creating computer music, although he also used and created
sounds using Cmix rather than Csound. I'm not sure if Csound has
resynthesizing magic that makes this not necessary.
my .02
--
Michael Coble, Time Inc. New Media, Pathfinder
Music: http://pathfinder.com/pathfinder/staff/mcoble/music/
Gallery: http://www.panix.com/~coble, representing various artists
Hitman: http://pathfinder.com/pathfinder/staff/mcoble/
Work: http://pathfinder.com/
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Date: Mon, 24 Aug 1998 22:34:14 -0700
From: Sean Costello
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To: jose halac
CC: csound@maths.ex.ac.uk
Subject: Re: winsound perf
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jose halac wrote:
>
> hi...
>
> Im trying to use winsound on a gateway 2000, windows based system.
>
> I am using word for orc/sco, and I'm using the toot.orc toot.sco tutorial
> files.
My guess is that MS Word is inserting all sorts of useless MS Junk into
your orchestra and score files. Try using Notepad for your orchestra
and score editing. There is also a Windows version of Emacs that would
be useful (this is what I use, after finding that Notepad put some
strange characters into some orchestra files).
To change the editor used by Winsound, go into the Windows directory on
your main drive, and open up WINSOUND.INI using Notepad. Where it says
OrchEdit and ScoreEdit, change those lines to indicate the pathway to
the editor you wish to use. For example, if you wish to use Notepad,
change these lines to
OrchEdit=C:\Windows\Notepad.exe
ScoreEdit=C:\Windows\Notepad.exe
This may be a really obvious thing to do, but it took me a while to
figure this out. Removing Csound and Winsound from the harddrive didn't
solve the problem, as the WINSOUND.INI file isn't located in the Csound
directory.
Hope this helps,
Sean Costelo
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Date: Mon, 24 Aug 1998 22:58:02 -0800
From: Mark Polishook
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I'm still looking to find a way to bring a table of 512 (or more values) into
Csound. As per a suggestion below:
Bertolt Sobolik wrote:
> Why not just write the values to a file and read them in with GEN 1?
I'm assuming that the table should look like
f1 0 512 -1 "voss" 0 4 0
and the file, called "voss," holds 512 values. But when using this snippet,
Csound complains that the table does not have a header:
Is there a way to bring this table into Csound?
Thanks,
-mp
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From: roger k
To: csound , jose halac
Subject: Re: winsound perf
Date: Tue, 25 Aug 1998 08:32:03 +0200
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>"sfinit=cannot open", and nothing happens.
Did you specify an outputfile?
--------
\ f(roger klaveness) = roger@apas.no +
\ http://home.c2i.net/roklaven/
--------
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From: David Schuyeteneer
To: Csound List
Subject: Belgian MP3 Server + WEB music broadcast
Date: Tue, 25 Aug 1998 11:54:20 +0200
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----------
> Van: hromlegn_kainn
> Aan: dBONANZAh@simsim.rug.ac.be; little.joe@simsim.rug.ac.be;
rammeke.pic@simsim.rug.ac.be
> Onderwerp: MP&G3_11h_650Mb
> Datum: maandag 24 augustus 1998 21:40
>
>
>
> Hello there,
>
> Maybe we talked about it before.
> It has taken some time to get properly organised.
> Finally it is there. We are starting up our web audio project.
> Read, enjoy & participate!
>
> The idea.
> Sola P., our dbonanzah! contact for South America,
> calculated that, with a decent MPEG3 compression
> we could make a cd containing 11 hours of music.
> We already had this ftp-server
> set up, so tht is what we are going to use,
> and we just hope that the connection can hold!
>
> The name.
> We have been mixing it up in all possible ways but it should all
> point in the same direction:
> MP&G_11H.dbh!
> mp3_11h_650Mb
>
> dBonanzaH!
> In reality, nothing coherent seemed to be happening here in
Belgium/Flanders,
> and we thought it was due to the will to set up an infrastructure
> where anti/multi/super/un - disciplinary digitaloz
> could connect to: audio & visio & action or just a big yell! we don't
care,
> as long as it is not Flemish, pragmatic and consolidating and ...
> still 100% digital! It is of no use limping after cultural moneymakers,
> kick start your own channels and get your feet wet!
> So the idea is to give space to an emerging electronic community
> of local artousiasts of any kind, provide them with tools for
> creativity and connect to similar initiatives world wide.
> Empowerment = digital!
>
> The task.
> Read the readme. Spread the news.
> Fwd it to interested noise/snd/music makers
> and please: upload YOUR music !!!
>
> You still don't know who we are and what we want you to do?
> Or what the hell is mpeg3 and how do I use it?
> All descriptions are at
> http://simsim.rug.ac.be/dBONANZAh/
> The readme.html/.txt is at
> ftp://quixot.rug.ac.be/MP&G_11h.dbh!/
>
> We are going to finish the program with an 11 hours webcast!
> Announced in due time. You can also contact our staff:
>
> dBONANZAh@simsim.rug.ac.be
> hromlegnkainn@simsim.rug.ac.be
> little.joe@simsim.rug.ac.be
> rammeke.pic@simsim.rug.ac.be
>
> And join the MP&G_11h.dbh! mailing list
> (with an empty email without subject to
> mp3_11h_650mb-subscribe@makelist.com)
>
> all the best,
> gz & fkk
>
> _._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._.
> [e r r o r = h o m e] http://simsim.rug.ac.be/staff/guy
> [anti_analog:hit_men] ftp://quixot.rug.ac.be *\HK/YFCF
> _._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._.
>
> advancements in technology have meant that all manner of equ
> ipment is now available for reappropriation by whoever has t
> he time to learn how to use, redefine, misuse and rewire it
> |