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Re: fractional part of p1

Date1999-04-19 18:48
FromJM&M DARREMONT
SubjectRe: fractional part of p1
It marks wich notes from a chord will be tied to wich notes of the next chord.
It allows pollyphonic ties.
Example:  i1.1 will be tied to only further i1.1 note.
You don't need it if your ties aren't pollyphonic.
There was a score  several weeks ago on this list demonstrating it.

Sergey Batov a écrit:
>
> "An optional fractional part can provide an additional tag for specifying
> ties between particular notes of consecutive clusters." Csound Manual.
> 
> can anybody explain how does fractional part of p1 works
> (what value must it be equals to, e.t.c.)?

-- 
Jean-Michel DARREMONT



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Date: Mon, 19 Apr 1999 13:54:21 -0400
From: nate drdul 
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hi, ive been lurking on the list for awhile but havent been able to get
much out
of csound untill reading the design tutorial from the first chapter of
the csound book.
Since hearing Terry Rileys Harp of New Albion ive been interested in
different tunings and just intonation.
My  question is what is the easiest way to go about exploring alternate
scales.
Im thinking it would be really interesting to have a set of conversation
functions like cpspch() for alternate tunings, or maybe a function that
would take in a scala file( the catalog of scala files is incredibly
dense) and could be used like
cpspch.
anyone got any ideas?
nate



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From: Patrick Robinson 
To: csound@maths.ex.ac.uk
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Subject: midi amplitude
Date: Mon, 19 Apr 1999 19:43:09 +0100
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Hello all,

I've been experimenting with using csound in realtime (Gabriel Maldonado's
version) and  a regular problem for me is regulating the amplitude of the
overall output when using more than one note at a time,  i.e. any new note
essentially generates a new instrument and the amplitudes of each are
summed together on output - this can lead to clipping.

My question is this:- Is there any way of allowing Csound to recognise how
many midi notes are being pressed at any one time?

thanks,

Pat




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Date: Mon, 19 Apr 1999 17:42:41 -0400
From: Patrick Pagano 
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Hi folks
pat the pest still trying to get a long delay.
I am running Visual Orchestra on a IBM cloney
I have been able to re-create and do almost all toots using it. But I
fall short trying to get a long(ambient delay). Is anyone out there
using VIsOrc that can help=patchwork is still a little difficult for me
right now,so I would like to stick with VisOrc for a while until I get
the basics down and for me delay is a basic.
Thanks to Pete Traub and Hans for helping but unfortunately I am needing
a little more hand holding
Thanks again
Pat Pagano



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From: Michael Gogins 
To: ndrdul@sgi.net, csound@maths.ex.ac.uk
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Subject: Re: tunings and csound
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All you need to do is create a function table with your desired tunings,
then you can use it is a lookup table to get the precise frequency nearest
in pitch to other notes. In other words, it acts as an adjustment table to
move notes to the exact desired pitch. If you do this, the input (the pitch
pfield of the note) can be in any unit easiest for you to work with -
pitch-class, scale degree, cycles per second, linear octave, or whatever.
You could even use one pfield for octave and another pfield for scale
degree.

-----Original Message-----
From: nate drdul 
To: csound@maths.ex.ac.uk 
Date: Monday, April 19, 1999 1:49 PM
Subject: tunings and csound


>hi, ive been lurking on the list for awhile but havent been able to get
>much out
>of csound untill reading the design tutorial from the first chapter of
>the csound book.
>Since hearing Terry Rileys Harp of New Albion ive been interested in
>different tunings and just intonation.
>My  question is what is the easiest way to go about exploring alternate
>scales.
>Im thinking it would be really interesting to have a set of conversation
>functions like cpspch() for alternate tunings, or maybe a function that
>would take in a scala file( the catalog of scala files is incredibly
>dense) and could be used like
>cpspch.
>anyone got any ideas?
>nate
>



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From: Hans Mikelson 
To: Csound 
Subject: Re: Popcorn fractal: Orc/Sco
Date: Mon, 19 Apr 1999 18:34:39 -0500
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>I wonder, does anyone know anything about the so-called "warm" or "analog"
>digital filters:


The best one in Csound now is probably moogvcf however when compared 1:1
with a real analog filter the analog sounds:

More vocal, more "up front", less muddy/mushy.  Not sure why but perhaps
phase plays a role?

Regards,
Hans Mikelson



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From: Hans Mikelson 
To: csound@maths.ex.ac.uk
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Subject: Re: Chebyshev polynomials (was: Re: Distorsion)
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>Now look at the table. It has a huge DC-offset and its zero crossing
>is not in the middle.

Try using only odd harmonics with values less than one and make sure the
input signal is less than 1 in amplitude.

This does not sound the greatest to my ears but here is an example:

; ORCHESTRA
sr     =    44100
kr     =    4410
ksmps  =    10
nchnls =    2
gaout  init 0

       instr    2

idur   =        p3
iamp   =        p4
ifqc   =        cpspch(p5)

aamp   linseg   0, .01, 1, idur-.03, 1, .01, 0, .01, 0
asig   pluck    iamp, ifqc, ifqc, 0, 1
aflt   butterlp asig, 8000
gaout  =        gaout + aflt*aamp

       endin

       instr    3

idur   =        p3
iamp   =        p4
ifqc   =        cpspch(p5)

aamp   linseg   0, .01, 1, idur-.02, 1, .005, 0, .005, 0
asig   oscil    iamp, ifqc, 1
gaout  =        gaout + asig*aamp

       endin

; Guitar Amp
       instr    11

idur   =        p3
ipre   =        p4
ipost  =        p5
ish1   =        p6
adel   init     0
achrs1 init     0
achrs2 init     0

asig   =        gaout/50000*ipre
adst   table3   asig, ish1, 1, .5
ascl   =        adst*50000

arat1  expseg   .2, idur, 8
alfo1  oscil    4, arat1, 1
achrs1 vdelay3  ascl+achrs1*.95, 10+alfo1, 50
alfo2  oscil    4, arat1, 1, .25
achrs2 vdelay3  ascl+achrs2*.95, 10+alfo2, 50

krms   rms      ascl, 100

apeq1  pareq    ascl,  100-krms/100, 1.2, .7, 0
apeq2  pareq    apeq1, 400-krms/20, .5, .7, 0
apeq   pareq    apeq2, 1500-krms/7, 1.5, .7, 0
aflt   butterlp apeq, 6000

avrbl1 nestedap  aflt*.5+adel*.2, 2, 1, .053, .1, .033, .12
avrbr1 nestedap  aflt*.5-adel*.2, 2, 1, .051, .1, .031, .12
avrbl2 nestedap  avrbl1-adel*.2, 3, 1, .253, .04, .133, .03, .071, .02
avrbr2 nestedap  avrbr1+adel*.2, 3, 1, .251, .04, .141, .03, .063, .02

avrbl  =         avrbl1+avrbl2
avrbr  =         avrbr1+avrbr2

adflt  butterlp  avrbl+avrbr, 500
adel   delay     adflt, .101

gaout  =        0
aoutl  butterhp aflt*ipost+avrbl*.8+achrs1*.5, 20
aoutr  butterhp aflt*ipost+avrbr*.8+achrs2*.5, 20
       outs     aoutl, aoutr

       endin

; SCORE
f1 0 8192 10 1

;   Sta  Dur  Amp   Pitch  Pre  Post  Shape1  Shape2
;i1  0    .5   10000 7.00   10   1     0       0
;i1  +    .    .     .      .    .     .1      .
;i1  .    .    .     .      .    .     .2      .
;i1  .    .    .     .      .    .     .1     -.1
;i1  .    .    .     .      .    .     .2     -.2

;   Sta  Dur   Amp   Pitch
i2  0    4.5   5000  6.02
i2  0    4.5   5000  7.06

;i2  1.25  .25   2000  7.07
;i2  +    .     3000  7.00
;i2  .    .     4000  8.00
;i2  .    .     5000  8.05
;i2  .    .     6000  9.00

;i2  2.75 .25   6000  7.07
;i2  +    .     7000  7.00
;i2  .    .     8000  8.00
;i2  .    .     9000  8.05
;i2  .    .    10000  9.00


;   Sta  Dur   Amp   Pitch
;i2  4.5  4.5   5000  5.10
;i2  4.5  4.5   5000  7.05

;i2  5.75 .25   2000  7.07
;i2  +    .     3000  7.00
;i2  .    .     4000  8.00
;i2  .    .     5000  8.05
;i2  .    .     6000  9.00

;i2  7.25 .25   6000  7.07
;i2  +    .     7000  7.00
;i2  .    .     8000  8.00
;i2  .    .     9000  8.05
;i2  .    .    10000  9.00

;           GEN Int Scaling  DC Fund 2nd 3rd  4th  5th
f11 0 65537 13  1   1        0  .5   0   .5  0    .5   0   0
;    Sta  Dur  Pre  Post  Table
i11  0    5.5   2    .3    11




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Date: Mon, 19 Apr 1999 16:49:59 -0700
From: Sean Costello 
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Hans Mikelson wrote:
> 
> >I wonder, does anyone know anything about the so-called "warm" or "analog"
> >digital filters:
> 
> The best one in Csound now is probably moogvcf however when compared 1:1
> with a real analog filter the analog sounds:
> 
> More vocal, more "up front", less muddy/mushy.  Not sure why but perhaps
> phase plays a role?

Probably has to do more with the nonlinearities in the circuit
(distortion from the differential pair, distortion between the stages of
the transistor ladder). These would be a bear to implement, because the
distortion from the differential pair, besides being a form of soft
limiting (hard enough to implement digitally), would form a part of the
feedback path in the circuit. Having nonlinearities in a feedback path
is a nice way to get aliasing, amplitude overflow, and other nasty
digital artifacts.  People like Harvey Thornburg at CCRMA are working on
developing bandlimited nonlinearities for just this very reason.

Anyway, the moogvcf opcode still sounds pretty good. Not really like a
Moog filter - more like the Roland SH-101, which has the same design as
the Moog filter as far as having 4 cascaded one-pole filters with an
overall feedback path, but uses an entirely different analog circuit
based on OTA's as opposed to a transistor ladder. It certainly sounds
more analog than the state-variable circuit I coded the other day, which
is useful, but sounds very brittle.

Probably the best way of emulating an analog synth with the existing
opcodes is by using several bandlimited pulse waveforms (gbuzz), mixed
together, run through some sort of nonlinear clipping function, then run
through the moogvcf. The nonlinear clipping function will simulate the
types of intermodulation distortion that will be generated in a Moog
when the input waveforms overdrive the differential input of the filter.
Synths like the Minimoog and Moog Rogue featured this type of distortion
as an integral part of their sound. Running the output of the moogvcf
through distortion would also be useful, as the VCA used in the Minimoog
also displayed nonlinear behavior.

Sean Costello


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To: csound@maths.ex.ac.uk
From: "A. Archias" 
Subject: Re;Chebyshev polynomials (was: Re: Distorsion)
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>Now look at the table. It has a huge DC-offset and its zero crossing
>is not in the middle.

>>>>
 As I understood it I thought pending on the coefficients you use
under the cheb polynomal you could easily end up with a net dc
offset therefore requiring a dc block or high pass function.
These coefficients are adding and subtracting in differing amounts
of power terms, (a3)*4x^3-3x +(a4)*8x^4-8x^2+1 + etc. so the running averages
can end up non-zero. This was in a previous thread.

 Andy Archias





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Date: Mon, 19 Apr 1999 20:18:09 -0700
From: Prent Rodgers 
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Nate,

I have been exploring Partch's 43 tone to the octave tonality diamond
using Csound for a few months now. I use a function table in my score
file like the following. I split it across several lines so it is easier
to read.

f3 0 64 -2 .0000000 .0021506 .0053273 .0084467 .0111731 .0150637
.0165004
.0182404 .0203910 .0231174 .0266871 .0294135 .0315641 .0347408 .0386314
.0417508
.0435084 .0470781 .0498045 .0519551 .0551318 .0582512 .0617488 .0648682
.0680449
.0701955 .0729219 .0764916 .0782492 .0813686 .0852592 .0884359 .0905865
.0933129
.0968826 .0996090 .1017596 .1034996 .1049363 .1088269 .1115533 .1146727
.1178494
.1200000

Then in the .orc file, I index into this table to select the proper
frequency like this:

ipitch   table   p5, 3            ; convert note number 1-43 to
oct.fract format
icps     =       cpspch(ioct+ipitch)  ; convert oct.fract to Hz

Then I can select any of the 43 tones with a note number (p5)  in the
score. I also have a separate octave number in the score, as recommended
by Michael Gogins.

For a complete picture of the orchestra and scores working together to
learn the tonality diamond, (shameless plug on again) see my web site:
http://pws.prserv.net/music1 or .mp3 example files at
http://www.mp3.com/PrentRodgers .

Prent Rodgers
Mercer Island, WA



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Subject: shareware sample editor
From: Nathan Day 
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There was a great shareware sample editor for the PowerMac, it was called
something like IWE and I think it was writen by akia for use with there
samplers but made publicly available. Does anybody no where I can find it or
even what it is called so I can do a search for it.

Nathan Day
nathand@senet.com.au


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http://www.sonicworx.com




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Date:     Tue, 20 Apr 99 12:24:24 BST
From: jpff@maths.bath.ac.uk
Subject:  Re: CSound Amiga Version
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Message written at 20 Apr 1999 09:41:07 +0100
--- Copy of mail to pipe@algonet.se ---
In-reply-to:  (message from Anders
	Andersson on Fri, 16 Apr 1999 00:00:34 +0200)
References:  

>>>>> "Anders" = Anders Andersson  writes:

 Anders> I'm very eager to see an official Amiga-dist of CSound in the future,
 Anders> as I now believe I'm capable of handling that project.

Great, as that was a processor missing from the collection

 Anders> But there are some things that I could need a bit help with before I
 Anders> start planning on a clean Amiga dist. I need to write graphic routines
 Anders> for AmigaOS, but I don't really know how CSound calls the different
 Anders> window routines. I have managed to get a whole lot of information
 Anders> by reading /winX11.c/ and /winbor.c/ etc etc, but there are still some
 Anders> things I don't really get..

 Anders> Are there any documentation on how to write graphic routines for CSound,
 Anders> or do I have to look at the different window-handlers that's already
 Anders> implemented?

Documentation no, but I have done a couple of implementations and I
would that it is not really very hard.  You need to write functions
int Graphable(void) /* called during program initialisation */
void MakeGraph(WINDAT *wdptr, char *name) 
	/* called from window.c to open a graphics window */
void DrawGraph(WINDAT *wdptr)    /* called from window.c to graph an array */
void KillGraph(WINDAT *wdptr) /* Destroy a graph */
int ExitGraph(void)           /* stop graphs tptally */

and if you want to do xyin opcode you need the following two; or at
least dummies liek these

void MakeXYin(XYINDAT *wdptr, float x, float y)	/* initial proportions */
{
}

void ReadXYin(XYINDAT *wdptr)
{
}

I suggest that you look at winfg.c and winbor.c as simple cases.

 Anders> I have another problem with the Amiga-version, that has to do with the
 Anders> compiler I currently use /(gcc)/.
 Anders> It seems that the function "/tell()/" isn't imlemented, so all the opcodes
 Anders> that use external samples don't work! =(.

I have removed tell from currect sources as it is not defined in all
systems.  In sysdep.h I have added
# define tell(fd) lseek(fd, 0, SEEK_CUR)
for some platforms at least.  I suggest that you do the same.  If you
want I can send you my current sources

 Anders> Strange thing though.. All the other Level 1 functions are implemented..

 Anders> Are there any major reason to use Level 1 files? Or could they be rewritten
 Anders> to use Level 2 (well, it's ANSI-C).
 Anders> I assume all the file-handling routines are very deep in CSound, and would
 Anders> be a *huge* pain to change, but if it's possible..

I have considered doing this rewrite, so it woudl be closer to real
ANSI, but always shirked away from the disruption.  I think it should
be done, but there are places where an int as a file descriptor is
rather deep, and so converting to a FILE* is a pain.  One could use
the fileno function, but that is not standard either.

 Anders> The last but not least question:
 Anders> Where can I find the absolute latest and recently updated
 Anders> CSound documentation?
 Anders> I've only found the online version, covering CSound V3.47.

In ftp.maths.bath.ac.uk:pub/dream/newest/ you can find both a ZIP and
a gzip-tar file of the 3.53 sources.  You can have 3.54 putative
sources from me.  

I have not decided on a time table for 3.54.  At present it is just
some random changes (drag'n'drop for Windows, optimising soem filters,
better use for linevents, one new opcode, bug fixes in the FM4
instruments, better handling of piped output and some minor changes.
Hardly worth a release yet.

=John ffitch