| >Please forgive if this is not interesting.
all u!ch = permeatez l!f = !nterezant + ver!tabl auss! +?
>The perl code below is a boiled down version
>of a program that wants to make a burst of notes
>which is dense at first, and then gets lighter.
>Much like the pattern that comes out of a spray
>paint can held at an angle to the surface.
>
>Braving the exposure of my stupidity, I am asking,
>why does the first piece of code produce linear
>results, while the second produces the desired effect?
>
>Thanks,
>
>Toby
>
>
>--------------------------------------------------------
>
>#!/usr/bin/perl
>
>for($x = 0; $x < 30; ++$x){
> $it = 100 * rand * rand * rand;
> print "It is $it\n";
>}
>
>print "XXXXXXXXXX\n";
>
>for($x = 0; $x < 30; ++$x){
>
> $it = 100;
> $it *= rand;
> $it *= rand;
> $it *= rand;
>
> print "It is $it\n";
>}
>
#!/usr/bin/perl
for($x = 0; $x < 30; ++$x){
$it = 100 * rand * rand * rand;
print "It is $it\n";
}
print "XXXXXXXXXX\n\n\n";
for($x = 0; $x < 30; ++$x){
$it = 100;
$it *= rand;
$it *= rand;
$it *= rand;
print "It is $it\n";
}
print "XXXXXXXXXX\n\n\n";
for($x = 0; $x < 30; ++$x){
$it = ((100 * rand) * rand) * rand;
print "It is $it\n";
}
rezultat. konztataz!on +?
It is 65.3782831970602
It is 49.2861914914101
It is 15.9146191552281
It is 71.64664901793
It is 63.5407107882202
It is 26.1497096624225
It is 58.3395312540233
It is 3.52130266837776
It is 98.4509238041937
It is 70.5827633384615
It is 71.2962868623435
It is 21.7319569084793
It is 33.1810324918479
It is 10.2283947635442
It is 41.3135713431984
It is 59.3739454168826
It is 30.2796812728047
It is 2.66586630605161
It is 11.8343100883067
It is 60.733166243881
It is 74.1944562643766
It is 30.8538358192891
It is 53.627393534407
It is 60.4929444845766
It is 40.6305307056755
It is 61.3988273311406
It is 62.5441311858594
It is 44.4350824225694
It is 82.6332832220942
It is 40.4510914348066
XXXXXXXXXX
It is 1.61255860887478
It is 0.962566716476907
It is 5.63174091814309
It is 3.20378403749047
It is 6.69740020113364
It is 7.38190725151666
It is 0.424975992108451
It is 0.78085950346134
It is 32.1432608180496
It is 5.14341727209989
It is 0.785035386632422
It is 0.459627419399763
It is 2.64281031315831
It is 16.4509422347959
It is 1.70579100510757
It is 30.3714638631455
It is 1.7551565025779
It is 2.16349358581452
It is 0.207139034450823
It is 47.988438945186
It is 0.945941457706158
It is 13.9242847560299
It is 25.1066254931667
It is 3.92672395627734
It is 0.138673795261964
It is 1.17430959974168
It is 0.743956403280891
It is 12.1777855269806
It is 38.1751602122164
It is 19.1047808118941
XXXXXXXXXX
It is 2.11494557823931
It is 29.1435927541792
It is 11.691548781336
It is 47.9766190735471
It is 22.8273503735739
It is 52.114046973826
It is 7.98152072725997
It is 0.598772082738917
It is 0.237762282237206
It is 6.11376474940015
It is 2.55184297982819
It is 5.75451573266233
It is 27.0852921207915
It is 3.61822598637296
It is 4.80687540222597
It is 23.6076253091751
It is 7.54673424365724
It is 0.348335303155861
It is 6.17136597042529
It is 0.128532694989869
It is 1.93330287147658
It is 13.0351859158493
It is 2.71011624084144
It is 6.03302952339297
It is 29.4178783838623
It is 27.2968478154298
It is 6.33425205969221
It is 1.54675818591073
It is 14.6849522869513
It is 1.6428388087701
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Date: Mon, 23 Aug 1999 11:45:17 -0400
From: Antoine Lefebvre
Subject: init error again
To: Mailing List
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I change kgoto by goto to avoid initialisation but I get the same error...
INIT ERROR: oscili: not initialized
here is the orc:
instr 1
aout init 0
asig in ; get the live input
krms rms asig
if krms > 700 goto startinst
goto end
startinst:
aout oscili 10000, 440, 1
end:
out aout
endin
--
Antoine Lefebvre
antoinelefebvre@softhome.net
http://pages.infinit.net/linux/rocket/index.html
http://pages.infinit.net/linux/music/music.html
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Date: Sat, 24 Jul 1999 09:50:20 -0700
From: Tobiah
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Subject: score.srt
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I wish that I could bypass recreating the score.srt when
it already exists, and is not out of date. Didn't somebody
say that there is a flag which will allow me to forgo the
generation of the score.srt and just use the existing one?
I can't find the flag.
It takes a while to sort a very large score.
Thanks,
Toby
~
~
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Date: Sun, 25 Jul 1999 01:37:43 +0100
From: de Jong
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Subject: feedback fm
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Hi,
Lately I've been playing around with an oscil which is being fm-ed by
itself.
like this:
a1 init 0
a1 oscili 1, kfreq+a1*(kfb*kfreq), 1
where kfb is the amount of feedback and sr = kr.
The problem is, when I increase the feedback the frequency goes down. So
I should probably increase kfreq by some factor if I increase kfb. But
how much? How could I calculate kfreq so that the resulting frequency
stays the same if I increase kfb?
Can anyone help?
thnaks,
Fokke de Jong.
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To: csound@maths.ex.ac.uk
Subject: Re: sequencer
In-reply-to: Your message of "Fri, 23 Jul 1999 19:04:23 EDT."
<000f01bed55f$b4ce9740$79d496c0@Realizer.ngt.sungard.com>
Date: Sat, 24 Jul 1999 09:11:19 -0400
From: Paul Barton-Davis
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Michael Gogins writes:
>MIDI note offs are not difficult to handle if you use a push-down stack of
>running instrument instances for each MIDI channel/MIDI key slot. Note on
>messages push an instrument instance onto the stack, and note off messages
>pop an instrument instance off the stack. This is not perfect, but it works
>well enough.
i think its wrong, however :) most "real" keyboard devices i use seem
to use FIFO ordering for note on/off, not LIFO. so you actually want a
queue, not a stack.
>In my view, the MIDI opcodes of Csound should be supplemented by an option
>to translate MIDI channel messages into score events and do the
>pushing/popping to handle note on/note off. Then people could develop,
>debug, and maintain one single instrument definition that would work for
>both realtime and nonrealtime synthesis.
lots of non-MIDI Csound opcodes are extremely also MIDI unfriendly, since
they require a note duration which is unavailable for live MIDI-driven
performance.
--p
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From: Tobiah
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Please forgive if this is not interesting.
The perl code below is a boiled down version
of a program that wants to make a burst of notes
which is dense at first, and then gets lighter.
Much like the pattern that comes out of a spray
paint can held at an angle to the surface.
Braving the exposure of my stupidity, I am asking,
why does the first piece of code produce linear
results, while the second produces the desired effect?
Thanks,
Toby
--------------------------------------------------------
#!/usr/bin/perl
for($x = 0; $x < 30; ++$x){
$it = 100 * rand * rand * rand;
print "It is $it\n";
}
print "XXXXXXXXXX\n";
for($x = 0; $x < 30; ++$x){
$it = 100;
$it *= rand;
$it *= rand;
$it *= rand;
print "It is $it\n";
}
-------------------------------------------------------
It is 14.4036914221942
It is 64.0403301455081
It is 92.0018395874649
It is 86.0313134733588
It is 94.1714564803988
It is 28.7829398643225
It is 4.56200079061091
It is 52.9926896095276
It is 12.277115136385
It is 65.0413279421628
It is 7.03556826338172
It is 58.5801591631025
It is 43.5281078331172
It is 64.0121327247471
It is 58.7811752222478
It is 24.898499250412
It is 30.1713058259338
It is 69.1552561242133
It is 43.1361183989793
It is 44.8441555257887
It is 6.85915164649487
It is 66.7636311613023
It is 56.5284450538456
It is 47.2567927092314
It is 76.6749755479395
It is 30.751428892836
It is 75.7148835342377
It is 22.4481104407459
It is 45.5015169922262
It is 30.3448522929102
XXXXXXXXXX
It is 0.76186388394069
It is 21.3483627110067
It is 34.4190002005938
It is 0.130956948373957
It is 0.239052579001631
It is 1.3333306923856
It is 16.7682933320958
It is 33.6714633776643
It is 25.1988224999129
It is 56.3710418350352
It is 0.614277701366594
It is 34.1737601536983
It is 0.581045287377612
It is 19.6711942362409
It is 4.92022134523882
It is 2.09921444154036
It is 3.04495186204821
It is 2.59705455484509
It is 37.7499712346738
It is 17.5143909592512
It is 14.5600427698617
It is 34.2413973866594
It is 31.0327469171314
It is 1.52027668544987
It is 2.2485724867584
It is 10.6191589151146
It is 0.827340811845958
It is 0.365990752095503
It is 6.01584646652215
It is 32.4042397744241
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Date: Sat, 24 Jul 1999 23:31:56 +1000
From: Bob Douglas
Reply-To: rdouglas@mail.usyd.edu.au
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I know it's OT, but does anyone have any ideas of how to
translate/import/migrate old Professional Composer notation files into
Finale or Sibelius ?
Any help would be greatly appreciated,
Bob Douglas
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Date: Mon, 23 Aug 1999 18:14:43 +0000
From: pete moss
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To: "csound@maths.ex.ac.uk"
Subject: Re: init error again
References: <19990823114517.A11131@tiger> <37C18F0D.2469B9C@bigfoot.com>
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[oops! meant to copy this to the list as well]
the oscili opcode is a bit of c code that needs to have an init function
run before it can work. i have had this same problem recently. the
quick answer is dont put oscils in goto statements because they wont be
initialized. you need to get more creative in your logic if you want
this to work.
try something like this:
instr 1
aout oscili 10000, 440, 1
asig in ; get the live input
krms rms asig
if krms <= 700 kgoto end
out aout
end:
endin
this way the oscil is always running and it gets its init correctly.
sound only gets output at certain times.
this example might need more tweaking to get the desired results. im
not sure is out has the same init requirements.
:P
>
> Antoine Lefebvre wrote:
> >
> > I change kgoto by goto to avoid initialisation but I get the same error...
> >
> > INIT ERROR: oscili: not initialized
> >
> > here is the orc:
> > instr 1
> > aout init 0
> > asig in ; get the live input
> > krms rms asig
> > if krms > 700 goto startinst
> > goto end
> > startinst:
> > aout oscili 10000, 440, 1
> > end:
> > out aout
> > endin
> > --
> > Antoine Lefebvre
> > antoinelefebvre@softhome.net
> > http://pages.infinit.net/linux/rocket/index.html
> > http://pages.infinit.net/linux/music/music.html
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Date: Mon, 23 Aug 1999 14:41:14 -0400 (EDT)
From: Michael Coble
To: Tobiah
cc: Csound List
Subject: Re: A bit off topic->score generation
In-Reply-To: <3799FD34.4C99C8B4@rcsreg.com>
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> Please forgive if this is not interesting.
>
> The perl code below is a boiled down version
> of a program that wants to make a burst of notes
> which is dense at first, and then gets lighter.
> Much like the pattern that comes out of a spray
> paint can held at an angle to the surface.
>
Hello,
This comes from perldoc "do a perldoc -f rand".
rand EXPR
rand Returns a random fractional number greater than or equal to
`0' and less than the value of EXPR. (EXPR should be
positive.) If EXPR is omitted, the value `1' is used.
Automatically calls `srand....
It says that rand will either give a random number between 0 and
"EXPR" or between 0 and 1 if "EXPR" isn't there. So the first example
is really more like: $it = (((100 * rand) * rand) * rand); and each
successive call is getting a random number between 0 and the result of
a call to rand. The second is multiplying $it with a random number
between 0 and 1.
hope this helps,
-*-*-*-*-*-*-*-*-*-*-*-*-*=*-*-*
Michael Coble: Director, Client Engineering, 24/7 Media Inc.
funstuff: http://www.panix.com/~coble/
-*-*-*-*-*-*-*-*-*-*-*-*-*=*-*-*
> #!/usr/bin/perl
>
> for($x = 0; $x < 30; ++$x){
> $it = 100 * rand * rand * rand;
> print "It is $it\n";
> }
>
> print "XXXXXXXXXX\n";
>
> for($x = 0; $x < 30; ++$x){
>
> $it = 100;
> $it *= rand;
> $it *= rand;
> $it *= rand;
>
> print "It is $it\n";
> }
>
> -------------------------------------------------------
>
> It is 14.4036914221942
> It is 64.0403301455081
> It is 92.0018395874649
> It is 86.0313134733588
> It is 94.1714564803988
> It is 28.7829398643225
> It is 4.56200079061091
> It is 52.9926896095276
> It is 12.277115136385
> It is 65.0413279421628
> It is 7.03556826338172
> It is 58.5801591631025
> It is 43.5281078331172
> It is 64.0121327247471
> It is 58.7811752222478
> It is 24.898499250412
> It is 30.1713058259338
> It is 69.1552561242133
> It is 43.1361183989793
> It is 44.8441555257887
> It is 6.85915164649487
> It is 66.7636311613023
> It is 56.5284450538456
> It is 47.2567927092314
> It is 76.6749755479395
> It is 30.751428892836
> It is 75.7148835342377
> It is 22.4481104407459
> It is 45.5015169922262
> It is 30.3448522929102
> XXXXXXXXXX
> It is 0.76186388394069
> It is 21.3483627110067
> It is 34.4190002005938
> It is 0.130956948373957
> It is 0.239052579001631
> It is 1.3333306923856
> It is 16.7682933320958
> It is 33.6714633776643
> It is 25.1988224999129
> It is 56.3710418350352
> It is 0.614277701366594
> It is 34.1737601536983
> It is 0.581045287377612
> It is 19.6711942362409
> It is 4.92022134523882
> It is 2.09921444154036
> It is 3.04495186204821
> It is 2.59705455484509
> It is 37.7499712346738
> It is 17.5143909592512
> It is 14.5600427698617
> It is 34.2413973866594
> It is 31.0327469171314
> It is 1.52027668544987
> It is 2.2485724867584
> It is 10.6191589151146
> It is 0.827340811845958
> It is 0.365990752095503
> It is 6.01584646652215
> It is 32.4042397744241
>
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Message-ID: <37C16ED8.A9A058EF@intercom.es>
Date: Mon, 23 Aug 1999 17:55:04 +0200
From: Josep M Comajuncosas
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To: antoinelefebvre@softhome.net, csound@maths.ex.ac.uk
Subject: Re: init error again
References: <19990823114517.A11131@tiger>
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Hmmm what about this:
> INIT ERROR: oscili: not initialized
>
> here is the orc:
> instr 1
> aout init 0 ;<--- remove it (unnecessary with the modific=
ation)
> asig in ; get the live input
> krms rms asig
igoto startinst: ;ADD THIS LINE TO THE CODE
>
> if krms > 700 Kgoto startinst;replace by kgoto again =
(it
> won=B4t be necessari at the i-pass).
> Kgoto end;replace by kgoto again (it won=B4t be neces=
sari at
> the i-pass).
> startinst:
> aout oscili 10000, 440, 1
> end:
> out aout
> endin
The idea is to make sure that, both for krms>700 and krms<700, you don=B4=
t
skip
the initialisation of aout=B4s oscili. Probably your signal starts with a
low
amplitude and it skips the initialisation (goto end:).
--
Josep M Comajuncosas
C/ Circumval.lacio 75 08790 Gelida - Penedes
Catalunya - SPAIN tel. 93 7792243
e-mail: gelida@intercom.es
ET Informatica de Sistemes
e-mail: jcomajuncosas@campus.uoc.es
http://members.tripod.com/csound/
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Date: Mon, 23 Aug 1999 21:20:02 +0200
From: Josep M Comajuncosas
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Subject: buggy pitch detector
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Hi,
trying to overcome the current malfunction of the pitch opcode, I wrote
a pitch detector based on zero crossings (it should have been ok with
the kind of signal I need it to work).
It is a crap, mainly because it can only set pitches multiple of kr of
course (I forgot that detail...), and also because sometimes it fails...
even with a sine wave.
Anyway I thought somebody could be interested in it, or maybe could try
to improve the thing (I don=B4t suggest that...), so here it is
sr =3D 44100
kr =3D 44100
ksmps =3D 1
instr 1
;Autotuner; Josep M Comajuncosas ; Aug=B499
;really puts your instrument out of tune ;-)
ifreq =3D cpspch(p4)
irand =3D birnd(.5)
kfreq init ifreq*(1+irand)
asig poscil3 10000,kfreq,1,.35
out asig
timout 0, .02, skip; skip transient or 0 initial (optional)
;TRACK
;prefilter
alpf tonex asig,ifreq*1.1,20
klpf init 1
klpf0 =3D klpf
klpf downsamp alpf
klpf =3D int(klpf)
;zero-cross detector
kcounter init 0
kflag init 0
kpitch init ifreq
kpitch0 init kpitch
;skip once reached intended pitch ?...
kflag1 =3D ((klpf =3D=3D 0) ? (kflag=3D0?1:0):kflag)
if (kflag1 =3D 0 && kflag =3D 1) goto succeed
goto failed
succeed:
kpitch divz kr,kcounter,ifreq
kcounter =3D 0
goto continue
failed:
kcounter =3D kcounter+1
continue:
;printk 1/ifreq,kpitch
kflag =3D kflag1
;autotune
kdf init 1.0005;the bigger the faster but could cause clicks & other
things ;-)
if ((kpitch < ifreq && kpitch0 < ifreq) || (kpitch > ifreq && kpitch0 >
ifreq)) goto nochange
kdf =3D sqrt(kdf)
nochange:
if kpitch
To: Josep M Comajuncosas
cc: csound@maths.ex.ac.uk
Subject: Re: buggy pitch detector
In-Reply-To: <37C19EE1.3AC66D60@intercom.es>
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ive had better luck with counting samples between maxima (above a
above threshold) rather than using zero crossings... just read your
description not your orch so i dont know exactly what you are doing...
-matt
On Mon, 23 Aug 1999, Josep M Comajuncosas wrote:
> Hi,
> trying to overcome the current malfunction of the pitch opcode, I wrote
> a pitch detector based on zero crossings (it should have been ok with
> the kind of signal I need it to work).
> It is a crap, mainly because it can only set pitches multiple of kr of
> course (I forgot that detail...), and also because sometimes it fails...
> even with a sine wave.
> Anyway I thought somebody could be interested in it, or maybe could try
> to improve the thing (I don=B4t suggest that...), so here it is
>=20
> sr =3D 44100
> kr =3D 44100
> ksmps =3D 1
>=20
> instr 1
> ;Autotuner; Josep M Comajuncosas ; Aug=B499
> ;really puts your instrument out of tune ;-)
> ifreq =3D cpspch(p4)
> irand =3D birnd(.5)
> kfreq init ifreq*(1+irand)
> asig poscil3 10000,kfreq,1,.35
> out asig
> timout 0, .02, skip; skip transient or 0 initial (optional)
>=20
> ;TRACK
> ;prefilter
> alpf tonex asig,ifreq*1.1,20
> klpf init 1
>=20
> klpf0 =3D klpf
> klpf downsamp alpf
> klpf =3D int(klpf)
>=20
>=20
> ;zero-cross detector
> kcounter init 0
> kflag init 0
> kpitch init ifreq
> kpitch0 init kpitch
> ;skip once reached intended pitch ?...
>=20
> kflag1 =3D ((klpf =3D=3D 0) ? (kflag=3D0?1:0):kflag)
> if (kflag1 =3D 0 && kflag =3D 1) goto succeed
> goto failed
> succeed:
> kpitch divz kr,kcounter,ifreq
> kcounter =3D 0
> goto continue
> failed:
> kcounter =3D kcounter+1
>=20
> continue:
> ;printk 1/ifreq,kpitch
> kflag =3D kflag1
>=20
> ;autotune
> kdf init 1.0005;the bigger the faster but could cause clicks & other
> things ;-)
>=20
> if ((kpitch < ifreq && kpitch0 < ifreq) || (kpitch > ifreq && kpitch0 >
> ifreq)) goto nochange
> kdf =3D sqrt(kdf)
>=20
> nochange:
>=20
> if kpitch lower:
> kfreq =3D kfreq/kdf
> goto next
> higher:
> kfreq =3D kfreq*kdf
> goto next
>=20
> next:
> kpitch0 =3D kpitch
> skip:
> endin
>=20
>=20
> f1 0 8192 10 1
> i1 0 3 7.00
> i1 + . 8.00
> i1 . . 9.00
> i1 . . 6.00
> i1 . . 10.00
> e
> --
> Josep M Comajuncosas
> C/ Circumval.lacio 75 08790 Gelida - Penedes
> Catalunya - SPAIN tel. 93 7792243
> e-mail: gelida@intercom.es
> ET Informatica de Sistemes
> e-mail: jcomajuncosas@campus.uoc.es
> http://members.tripod.com/csound/
>=20
>=20
>=20
>=20
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Date: Mon, 23 Aug 1999 22:32:12 +0100
From: Richard Dobson
Organization: Composers Desktop Project
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To: "csound@maths.ex.ac.uk"
Subject: Re: init error again
References: <19990823114517.A11131@tiger> <37C18F0D.2469B9C@bigfoot.com> <37C18F93.C3F6F655@bigfoot.com>
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Just a little question - unless it is actually required to restart the
oscillator at zero phase, how about just controlling it's envelope, or
even multiplying its output by zero?
Richard Dobson
pete moss wrote:
>
> [oops! meant to copy this to the list as well]
>
> the oscili opcode is a bit of c code that needs to have an init function
> run before it can work. i have had this same problem recently. the
> quick answer is dont put oscils in goto statements because they wont be
> initialized. you need to get more creative in your logic if you want
> this to work.
>
> try something like this:
> instr 1
> aout oscili 10000, 440, 1
> asig in ; get the live input
> krms rms asig
> if krms <= 700 kgoto end
> out aout
> end:
> endin
>
--
Test your DAW with my Soundcard Attrition Page!
http://wkweb5.cableinet.co.uk/rwd (LU: 6th July 1999)
CDP: http://www.bath.ac.uk/~masjpf/CDP/CDP.htm (LU: 14th June 1999)
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Date: Mon, 23 Aug 1999 21:40:20 +0000
From: pete moss
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Subject: eliminating crosstalk
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anyone know how to eliminate crosstalk? i have seen some talk about it,
but no concrete methods. anyone have any code examples or
descriptions? also, if i eliminate crosstalk in a signal and then play
the signal over two loudspeakers, does this damage the sound?
:P
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From: Hans Mikelson
To: CSound list
Subject: Re: nestedap questions
Date: Mon, 23 Aug 1999 18:13:38 -0500
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>I have played around with the nestedap opcode - it seems to exept its idel
in
>sec (not given in doc...). In this case the example in the doc is a little
>strange: more then 100 sec del time for a 4 sec note.
Ahh, yes, well, I meant to use milliseconds when I wrote the opcode but I
must have forgotten to divide by 1000 so the units are in seconds. The two
inner delay loops must fit inside of the outer delay loop so delay1 must be
longer than delay2 + delay3 times. I think it may have a problem if you
make delay1 equal to delay2 + delay3.
> imax = 1
> idel1 = .006
> igain1 = .3
> idel2 = .003
> igain2 = .3
> idel3 = .001
> igain3 = .3
>
> aout nestedap gasig, 3, imax, idel1, igain1, idel2, igain2, idel3,
igain3
>
> outs aout-gasig, aout+gasig
> endin
>
>Any idea?
See above.
Good luck,
Hans Mikelson
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From: Michael Gogins
To: mj@exbang.com, csound list
MMDF-Warning: Parse error in original version of preceding line at UK.AC.Bath.maths.omphalos
Cc: saol-dev list
Subject: RE: Csound orc to C converter ?
Date: Mon, 23 Aug 1999 19:12:54 -0400
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What would be the purpose of a Csound .orc to C compiler? I doubt it would
run any faster than Csound. Of course, the compiled programs would be able
to run by themselves - is that the idea?
-----Original Message-----
From: Michel Jullian [mailto:mj@exbang.com]
Sent: Wednesday, August 18, 1999 10:21 AM
To: csound list
Cc: saol-dev list
Subject: Csound orc to C converter ?
Hi,
Does anybody know of a Csound .orc to C converter (like there is SFront for
SAOL orchestras), either existing or in the works ?
--
Greetings,
Michel
.........................................................................
Michel Jullian Directeur General email mj@exbang.com
Exbang Industries S.A.
Mas Chauvain route de Villeneuve tel +33(0) 499 529 878
Maurin 34970 Lattes France fax +33(0) 499 529 879
.........................................................................
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From: Michael Gogins
To: jpff@maths.bath.ac.uk, csound@maths.ex.ac.uk
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Subject: RE: Csound on Windows98
Date: Mon, 23 Aug 1999 20:03:31 -0400
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I would imagine that the original message referred to AXCsound, not Csound.
An earlier version of AXCsound did require ML3211.dll. The current version
does not require it.
-----Original Message-----
From: owner-csound-outgoing@maths.ex.ac.uk
[mailto:owner-csound-outgoing@maths.ex.ac.uk]On Behalf Of
jpff@maths.bath.ac.uk
Sent: Saturday, August 21, 1999 11:38 AM
To: csound@maths.ex.ac.uk
Subject: Csound on Windows98
Message written at 21 Aug 1999 10:13:38 +0100
I received a message (by a round-about route) that a potential user of
Csound on Windows98 claimed that he needed a missing file, ML3211.dll,
in order to run Csound. I have neither W98 nor that file. Does
anyone recognise this DLL, or can confirm running/non running on W98?
Thanks for any insight
==John ffitch
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From: Michael Gogins
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Subject: RE: Csound on Windows98
Date: Mon, 23 Aug 1999 20:03:31 -0400
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I would imagine that the original message referred to AXCsound, not Csound.
An earlier version of AXCsound did require ML3211.dll. The current version
does not require it.
-----Original Message-----
From: owner-csound-outgoing@maths.ex.ac.uk
[mailto:owner-csound-outgoing@maths.ex.ac.uk]On Behalf Of
jpff@maths.bath.ac.uk
Sent: Saturday, August 21, 1999 11:38 AM
To: csound@maths.ex.ac.uk
Subject: Csound on Windows98
Message written at 21 Aug 1999 10:13:38 +0100
I received a message (by a round-about route) that a potential user of
Csound on Windows98 claimed that he needed a missing file, ML3211.dll,
in order to run Csound. I have neither W98 nor that file. Does
anyone recognise this DLL, or can confirm running/non running on W98?
Thanks for any insight
==John ffitch
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From: Michael Gogins
To: Lars Luthman , Csound list
Subject: RE: Unified file format
Date: Mon, 23 Aug 1999 20:13:49 -0400
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The implementation of csd files in AXCsound should be able to do what you
were trying to do. The code in the canonical sources does not skip over tags
until it finds , read tags until it finds
, and then skip all remaining tags, as it really should.
Instead, the canonical sources assume that the very first tag encountered
will be . This prevents the canonical Csound from
rendering a csd file embedded in another HTML or XML file. But, as I said, I
think AXCsound can do this - at least, it is SUPPOSED to do this.
-----Original Message-----
From: owner-csound-outgoing@maths.ex.ac.uk
[mailto:owner-csound-outgoing@maths.ex.ac.uk]On Behalf Of Lars Luthman
Sent: Monday, August 23, 1999 6:25 AM
To: Csound list
Subject: Unified file format
I sent this message yesterday, but I haven't seen it on the list, so here
it is again.
--
I was looking at some Csound compositions in unified file format (.csd),
and I saw that they had start and end tags that looked just like HTML. I
thought, "Cool! Does that mean that I can put a Csound composition in a
HTML file and maybe write some words about the composition, so the same
file can be read as a webpage in my browser and given to Csound for
rendering? That would be very convenient."
I copied a simple CSD file into the body of a HTML file and tried rendering
it with Csound (I changed the file name extension to .csd). It didn't work,
of course (as I would have known if I had read the manual). Perf complained
"Unknown command: " (or something like that, it froze before
finishing the line). Strange, it didn't say anything about "" being
an unknown command, and that line appeared before "". I tried moving
the contents of the CSD file almost to the beginning of the file, between
the "" and the "" tags, and this time it was rendered without
any problems.
Later, when I read the manual, I found out that it actually shouldn't have
worked the second time either, since the first line in a CSD file must be
"" and the last line must be "".
But I still think it would be a great if I could put a Csound composition
in a webpage, and don't have to copy and paste the composition itself to a
new textfile when I want to render it. So I have a suggestion.
Couldn't Csound ignore everything outside the "" tags?
That way a composition could be incorporated in any kind of file (e.g.
HTML, ASCII text, Word or PDF documents). Maybe someone could write a web
browser plugin that played Csound compositions!
What do you think? I'm afraid I don't have the time or the programming
skills to do any of these changes, but I sure would use it if it was done.
--ll
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From: prodgers@ibm.net
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To: csound list
Subject: Experimental Diamond Marimba web site using Java
Date: Mon, 23 Aug 1999 17:31:19 -0700
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Csounders:
I have been experimenting with Java applets in web pages recently, and have
created a web page with a Diamond Marimba on it. When it works, you can run
your mouse over the bars of the marimba and play the tonality diamond. Of
course, it doesn't always work...
The web page is off my home page at http://pws.prserv.net/music1 . The
applet depends on Java 1.2.2 and Swing. No commercial browser supports this
today. To see the applet, you have to download and install the SUN java
plug-in, which is a bit of a bear to get installed. Instructions are on the
web page.
The goal of the applet is to use a tonality diamond as the input to a
composition tool. I've got a bit more to go to do this, but perhaps
someday...
If the effort to support the SUN plug-in is too much, you can see a screen
shot of the applet on my home page.
Prent Rodgers
Mercer Island, WA
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From: John Starrett
To: prodgers@ibm.net
cc: csound list
Subject: Re: Experimental Diamond Marimba web site using Java
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Prent-
Your tonality diamond applet works great, and the plugin set up
easily. Congrats.
John Starrett
http://www-math.cudenver.edu/~jstarret
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nato.0+55. kontra. zte!m.lrg.pup!la
zcene.0+2. akt.0+0. \/\ m9ndzmk=m2c#tfr3!
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http://www.m9ndfukc.org/data/noisz/nato.0+55.kontra.ste!m.lrg.pup!le.mov
http://www.m9ndfukc.org/data/noisz/nato.0+55.kontra.ste!m.lrg.pup!le.0+2.mov
- la pup!le ezt maron ma!z __... http://m9ndfukc.com/b98c99
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From: Josep M Comajuncosas
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To: "Matt J. Ingalls"
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Subject: Re: buggy pitch detector
References:
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"Matt J. Ingalls" escribi=F3:
> ive had better luck with counting samples between maxima (above a
> above threshold) rather than using zero crossings... just read your
> description not your orch so i dont know exactly what you are doing...
Ok I see the instrument *can* indeed be improved a lot, first of all dete=
cting
only positive or negative zero crossings, eventually detect different not=
able
points (maxima, minima, +&- zero crossings) and weight the results, then =
get
the fractional part with some kind of movable mean. Also, discard too sho=
rt
periods based on a first estimated pitch.
Any other idea? ... I=B4ll improve the tracker.
--
Josep M Comajuncosas
C/ Circumval.lacio 75 08790 Gelida - Penedes
Catalunya - SPAIN tel. 93 7792243
e-mail: gelida@intercom.es
ET Informatica de Sistemes
e-mail: jcomajuncosas@campus.uoc.es
http://members.tripod.com/csound/
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Subject: Re: schedule, schedwhen - bug report
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I am also working with schedwhen and I was about to mail some questions to the
list when they were answered. I did find some ways to work around the bugs in
the code.
To have multiple activations of schedwhen, it needs to be reinited.
For instance:
initsched:
schedwhen ktrigger,.......
rireturn
if ktrigger !=0 goto continue
reinit initsched
continue:
The if-statement prevents schedwhen from reiniting and activating every
performance-pass for as long as ktrigger!=0. Once the trigger is back to 0, the
code is reinited and can be activated again.
It would be nicer, I think, if there was a release parameter in the opcode, more
like a drum-pad. But since there isn't, it makes sense to me to have to reinit
it for multiple activations.
To prevent the called instrument from stopping when the calling instrument
stops, the following helps, (I don't know why):
In the beginning of the instrument
itime times
p2=itime
On the other hand, to have it stop sooner, something like this works:
ktime timeinsts
if ktie>=p3 goto off
....
...
goto end
off:
turnoff
end:
I'm using these to write a kind of "a-live" sound-bugs that crawl around the
sound-stage, reproducing when they meet. It makes more sense to program the
algorithm in the instrument,in this case, than to use a score generating
software.
Gerard van Dongen gml@xs4all.nl
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From: richard bowers
To: csound csound
Subject: Re: schedule, schedwhen - bug report
Date: Tue, 24 Aug 1999 10:08:31 +0100
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> Many thanks, Rasmus.
>
> I'd been trying things out before I received your reply and some of what
you
> say has been experienced by me.
>
> One fudge, which seemed to work, for the problem of schedwhen switching
off
> the scheduled instrument when its own instrument ends is to trigger an
> intermediary instrument which contains schedule. I don't have the orch
with
> me but here is a description.
>
> ie.
> instr 1
> schedwhen "intrs 2 when something happens"
> endin
>
> instr 2
> schedule "instr 3"
> endin
>
> instr 3
> "carry out the scheduled event"
> endin
>
> I've used this in realtime to trigger an event when a signal crosses a
> threshold. It seems to work, but I'm not sure about what is going on
> internally, so to speak.
>
> Thanks for your advice. I do hope that the problems can be overcome and
the
> functionality extended to enable multiple triggering of schedwhen.
>
> Cheers,
> Richard Bowers.
>
> > richard bowers wrote:
> > >
> > > I understand that the opcodes schedule and schedwhen will only operate
> once
> > > in a single call to the instrument containing them. Is that correct?
> >
> > Yes. schedule only works at i-time and schedwhen has an internal
> > switch that is set to "off" after generating the event.
> >
> > Note that these opcodes are rather buggy:
> > First, they can not create deferred events (by setting i/kwhen > 0),
> > so the events are always started immediately. Any attempt to defer
> > events will give them wrong ringtime, namely i/kwhen + i/kdur, while
> > the turnoff time will be correctly set to start time+dur, which means
> > the instr will be around (at least) from generation time through to
> > deferred time + dur (ie def.time + p3).
> >
> > (Programmers note: As far as I can determine, there is no simple way
> > of creating deferred events from an opcode context in Csound, so I
> > wouldn't care to try to fix this feature, rather disallow it until
> > the rest is working.)
> >
> > Secondly, the generated events will not be turned off correctly,
> > but be left hanging, unless there has been some "real" score
> > (or midi or line input) event in the meantime.
> >
> > Prognotes: This is a complication in any Csound event generation from
> > an opcode. Event turnoff is not handled by kperf() (k-rate
performance),
> > but by playevents() (the score event scheduler, which calls kperf()
> > to run until next score event). kperf() returns only for new events
> > or for event turnoff occasions, but may be expired early by "realtime"
> > (midi, midi file or line input) events. Search for the variable
> > "sensType" in sources if you want to see how this works.
> >
> > Both sched* ugens call schedofftim() to set the turnoff time for
> > the generated event. The time is indeed set, but this is no help since
> > kperf() as noted doesn't check for turnoff. It does happen to work
> > for schedule since this opcode generates events at i-time, and so
> > will have the turnoff logged in playevents() for free before the first
> > k-cycle of the calling instr.
> >
> > My solution to this problem (yes, I've been working on and off for the
> > last few weeks to create an opcode to generate events at k-rate)
> > is to define a new "realtime" event, and do some changes to playevents()
> > and kperf() to have it handled. Not tested enough for publishing yet,
> > as I for one don't think that the compiler does all necessary QA...
> >
> > Thirdly, since they were created by copying a fair chunk of Csound core
> > code (the insert() func.) and doing some changes to that (rather than
> > setting up some scheme which involves calling this function), they are
> > (and will keep getting) out of synch with changes in that code,
> specifically,
> > I believe recent stuff like cpuprc and maxalloc will not work correctly
> > for instr's that have had events generated by sched* ugens.
> >
> > Finally, schedwhen shifts the arguments so that p4= p3, p5= intended p4
> etc
> > in the generated instrument. This one is trivial to fix (it has already
> > been done for schedule at one time).
> >
> >
> > In conclusion, besides wondering what the point of these opcodes is,
> > I'm curious as to whether they were tested _at_all_ before publishing?
> > (Usually I don't complain about stuff without trying to fix it,
> > but after looking at them I simply don't like them, and think they
> > should be taken out and sh (*ahem* deprecated and deleted)).
> >
> > Demonstration: The below csd file should generate one 2.33 second beep
> > after 0.5 seconds, and one after 5 seconds (modulus k-precision).
> > Just see what you get...
> >
> > Regards,
> >
> > re
> >
> >
> > ----------------------
> > schedbug.csd - demo some bugs in sched* opcodes
> > ----------------------
> >
> >
> > ;
> >
> > ;
> >
> > sr = 22050
> > kr = 441
> > ksmps = 50
> > nchnls = 1
> >
> > instr 1
> > ; Start i13 after .5 sec (not)
> > schedule 13, .5, 2.33, 550, 0
> > endin
> >
> > instr 2
> > ; Start i3 after 1 sec
> > ; Since this doesn't work, it starts after 4.0 instead of 5.0 secs
> > ; It should be turned off after 2.33 secs, but hangs until next score
> event
> > schedwhen 1, 13, 1, 2.33, 440
> > endin
> >
> > instr 13
> > ; itime is actual start time, p2 is the intended,
> > ; and p4=p3, p5=p4 if started by schedwhen
> > itime times
> > print itime, p2, p3, p4, p5
> >
> > ; Since p4 won't always be correct, we set a start pitch here
> > kpitch line 440, p3, p4
> > ; Now we're naughty to rub in the point: this linseg continues beyond p3
> > ; - but the error shows even if you remove the last ", .5, 20000" args
> > kamp linseg 25000, p3-.1, 25000, .1, 0, .5, 20000
> > ar oscili kamp, kpitch, 1
> > out ar
> > endin
> >
> >
> >
> > f1 0 4096 10 1
> >
> > i1 0 3.5
> > i2 4 6
> > f0 10
> >
> > e
> >
> >
> >
> >
>
>
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Date: Tue, 24 Aug 1999 12:54:19 +0200
From: Josep M Comajuncosas
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To: guto@sercomtel.com.br, csound@maths.ex.ac.uk
Subject: Re: Insisting on PITCH
References: <37C03B22.1796A79@intercom.es> <37C172A6.6EF@sercomtel.com.br> <37C1810A.F4379F4B@intercom.es> <37C1E285.6EFD@sercomtel.com.br>
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I wonder why those useful opcodes have been removed from the sources.
guto escribi=F3:
> DirectCsound 2.801 (windows...) still have spectrum and specptrk.
--
Josep M Comajuncosas
C/ Circumval.lacio 75 08790 Gelida - Penedes
Catalunya - SPAIN tel. 93 7792243
e-mail: gelida@intercom.es
ET Informatica de Sistemes
e-mail: jcomajuncosas@campus.uoc.es
http://members.tripod.com/csound/
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Date: Tue, 24 Aug 1999 09:51:06 -0600
To: csound@maths.ex.ac.uk
From: "17.hzV.tRL.478"
Subject: Re: detectbuggypittch.tch.tch.tchaschka.tchaj.
Sender: owner-csound-outgoing@maths.ex.ac.uk
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gelida@intercom.es ezkr!z
>"Matt J. Ingalls" escribi=F3:
>
>> ive had better luck with counting samples between maxima (above a
>> above threshold) rather than using zero crossings... just read your
>> description not your orch so i dont know exactly what you are doing...
>
>Ok I see the instrument *can* indeed be improved a lot, first of all detect=
ing
>only positive or negative zero crossings, eventually detect different notab=
le
>points (maxima, minima, +&- zero crossings) and weight the results, then ge=
t
>the fractional part with some kind of movable mean. Also, discard too short
>periods based on a first estimated pitch.
>Any other idea?
!mprov!ng odrz =3D 1eczusz !2 !mprov zelv +?
> ... I=B4ll improve the tracker.
Life is like a landscape. You live in the midst of it, but can describe it
only from the vantage point of distance. -Charles A. Lindbergh
a landscape Life is like. in the midst of it live You but can describe
from the vantage of distance point only it
Life is Lifelike can You describe only the midst of it
Ok I see the instrument indeed. *can* be improved a lot.
szuposz du =3D dez!r 2 add thought aftr thought 2 ur m9ndkonta!nr unt!l !t =
=3D
f!lz +?
dze prozesz =3D mozt klearl! undrztood !f du =3D ztart u!th nodz!ng
- first estimated pitch. =3D1 1
!t =3D reku!rz 1 !nf!n!t amnt ov pakag!ng 2 ztore nodz!ng
-
-m9ndfukc.com
17.hzV.tRL.478
e
|
| +----------
| | <
\\----------------+ | n2t=10
| >
e
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Date: Tue, 24 Aug 1999 12:35:28 -0400
From: "Job M. van Zuijlen"
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Organization: electona
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To: Josep M Comajuncosas
CC: csound@maths.ex.ac.uk
Subject: Re: Insisting on PITCH
References: <37C03B22.1796A79@intercom.es>
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I have had reasonable success with the pitch tracking using LPC
analysis. I don't know what is used for the pitch tracking, and whether
it is any different from pitch. While working with LPC I noticed that a
lot depends on the quality/clarity of the input signal, sloppy speech
will result in garbage, basically.
As you may discern from the failure of early guitar synthesizers, pitch
tracking is a complicated problem. For example, the fundamental
frequency of a 100 Hz tone may be absent, but because of the harmonics
(e.g., 200, 300, 400 Hz, etc.), you will still hear a 100 Hz tone,
because the periodicity hasn't changed. The waveform may change because
of spectrum changes, but the pitch may still be the same. So if you
want to compare one period with the next (I think you would use
something called autocorrelation for that) you have to take small or
even fairly sudden changes in the waveform into account. Not to
disillusion you, but don't feel too bad if you haven't solved this
problem in one afternoon.
Job van Zuijlen
Josep M Comajuncosas wrote:
>=20
> I=B4m sorry to insist on the topic, but... does anyone has a working
> example of a pitchtracker using the PITCH opcode, or anything else? I=B4=
m
> really needing it!
> Tnx
>=20
> --
> Josep M Comajuncosas
> C/ Circumval.lacio 75 08790 Gelida - Penedes
> Catalunya - SPAIN tel. 93 7792243
> e-mail: gelida@intercom.es
> ET Informatica de Sistemes
> e-mail: jcomajuncosas@campus.uoc.es
> http://members.tripod.com/csound/
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In-reply-to: <37C279DB.8EFD77E7@intercom.es> (message from Josep M
Comajuncosas on Tue, 24 Aug 1999 12:54:19 +0200)
Subject: Re: Insisting on PITCH
BCC: jpff@maths.bath.ac.uk
References: <37C03B22.1796A79@intercom.es> <37C172A6.6EF@sercomtel.com.br> <37C1810A.F4379F4B@intercom.es> <37C1E285.6EFD@sercomtel.com.br> <37C279DB.8EFD77E7@intercom.es>
Date: Tue, 24 Aug 99 18:14:56 BST
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Who removed spectrum and its friends? Which sources?
==John
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In-reply-to: <37C279DB.8EFD77E7@intercom.es> (message from Josep M
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Subject: Re: Insisting on PITCH
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Who removed spectrum and its friends? Which sources?
==John
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To: Csound list
From: Lars Luthman
Subject: Re: Unified file format
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>The implementation of csd files in AXCsound should be able to do what you
>were trying to do. The code in the canonical sources does not skip over tags
>until it finds , read tags until it finds
>, and then skip all remaining tags, as it really should.
>Instead, the canonical sources assume that the very first tag encountered
>will be . This prevents the canonical Csound from
>rendering a csd file embedded in another HTML or XML file. But, as I said, I
>think AXCsound can do this - at least, it is SUPPOSED to do this.
>
>
Sounds great, does it run on Mac? And if it does, where can I find it?
--ll
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To: Csound list
From: Lars Luthman
Subject: Looking for drums
Sender: owner-csound-outgoing@maths.ex.ac.uk
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I'm looking for bass drums (Csound instruments, not samples). I'm planning
to rewrite some of my XM files to Csound files, and I need some really
punchy bass drums (I'm mostly writing trance/techno stuff). I've
experimented with filtered noise and different kinds of envelopes, but I
just can't get it to sound good enough. Something that sounds similar to a
Roland TR-909 would be perfect.
I'm also looking for a 3.56 manual in Macintosh DOCMaker standalone
document format. I have the manual for version 3.47 in that format and I
really like it, it's much faster than starting Acrobat Reader or Navigator
on my computer. But I can't remember where I got it. Is there a manual in
that format for version 3.56?
-ll
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Message-ID: <37C2EB6E.AAF59B4C@ulster.net>
Date: Tue, 24 Aug 1999 14:58:54 -0400
From: Paul Winkler
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To: Lars Luthman
CC: Csound list
Subject: Re: Looking for drums
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Lars Luthman wrote:
>
> I'm looking for bass drums (Csound instruments, not samples).
There's a nice web page on getting drum sounds out of analog synths. The
theories should be applicable to Csound if you can work out the .orcs...
Somebody on the list sent me these links:
http://listen.to/concussor
http://www.zone-zero.com/as
http://www.users.globalnet.co.uk/~concuss
Here's some rough attempts I made based on that document:
; ORC
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;copyright 1998 Paul M. Winkler, zarmzarm@hotmail.com
;****++++
;**** Last modified: Wed Jul 21 07:05:31 1999
;****----
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
sr = 44100
kr = 22050
ksmps = 2
nchnls = 1
; Attempts at simulating analog drum synths.
instr 1 ; DUMB DRUM 1 (kick/tom)
; SELF-RESONATING LOWPASS FILTER
idur = p3
icutoff = p4
ires = 3 ; self-resonates at 1
iscale = 100000 / (ires) ; Not accurate, but
ballpark.
; Note that larger resonances make sound louder, richer, and fatter.
; Suggested range: 1 to 3 for deep stuff, up to 200 for very synthy.
; Try as low as .7 for short-decay, non-oscillating thuds, but then
amplitude must
; be seriously boosted.
asig osciln 1, 1/idur, 1, 1
; should use a better pulse signal, it's too "ticky".
kcutoff linseg icutoff, idur, 0 ; env. for filter
aout moogvcf asig, kcutoff, ires
out aout * iscale
endin
instr 2 ; DUMB DRUM 2
(kick/bass)
; SINE OSCILLATOR
idur = p3
ipitch = p4
iamp = p5
kenv linseg iamp, idur, .01 ; change to linseg to
get
; a total drum-n-bass bass sound. With expseg, decay of at
least
; 0.5 is nice; with linseg, as low as .1
asig oscili kenv, ipitch, 2
out asig
endin
;SCO
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;copyright 1998 Paul M. Winkler, zarmzarm@hotmail.com
;****++++
;**** Last modified: Tue Aug 24 14:39:34 1999
;****----
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
f1 0 1024 2 1 0 ; impulse... works but sounds
bad
f2 0 32769 10 1 ; sine wave
t 0 80
i1 0 .25 100
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . 50
s
t 0 100
i2 2 .1 100 30000
i2 3 . > .
i2 4 . > .
i2 5 . > .
i2 6 . > .
i2 7 . > .
i2 8 . > .
i2 9 . > .
i2 10 . > .
i2 11 . > .
i2 12 . 30 .
--
---------------- paul winkler ------------------
slinkP arts: music, sound, illustration, design, etc.
zarmzarm@hotmail.com --or-- slinkp AT ulster DOT net
http://www.ulster.net/~abigoo/
======================================================
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Date: Tue, 24 Aug 1999 14:57:04 -0400
From: Paul Winkler
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To: Lars Luthman
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Subject: Re: Looking for drums
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Lars Luthman wrote:
>
> I'm looking for bass drums (Csound instruments, not samples).
There's a nice web page on getting drum sounds out of analog synths. The
theories should be applicable to Csound if you can work out the .orcs...
Somebody on the list sent me these links:
http://listen.to/concussor
http://www.zone-zero.com/as
http://www.users.globalnet.co.uk/~concuss
Here's some attempts I made based on that document:
; ORC
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;copyright 1998 Paul M. Winkler, zarmzarm@hotmail.com
;****++++
;**** Last modified: Wed Jul 21 07:05:31 1999
;****----
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
sr = 44100
kr = 22050
ksmps = 2
nchnls = 1
; Attempts at simulating analog drum synths.
instr 1 ; DUMB DRUM 1 (kick/tom)
; SELF-RESONATING LOWPASS FILTER
idur = p3
icutoff = p4
ires = 3 ; self-resonates at 1
iscale = 100000 / (ires) ; Not accurate, but
ballpark.
; Note that larger resonances make sound louder, richer, and fatter.
; Suggested range: 1 to 3 for deep stuff, up to 200 for very synthy.
; Try as low as .7 for short-decay, non-oscillating thuds, but then
amplitude must
; be seriously boosted.
asig osciln 1, 1/idur, 1, 1
; should use a better pulse signal, it's too "ticky".
kcutoff linseg icutoff, idur, 0 ; env. for filter
aout moogvcf asig, kcutoff, ires
out aout * iscale
endin
instr 2 ; DUMB DRUM 2
(kick/bass)
; SINE OSCILLATOR
idur = p3
ipitch = p4
iamp = p5
kenv linseg iamp, idur, .01 ; change to linseg to
get
; a total drum-n-bass bass sound. With expseg, decay of at
least
; 0.5 is nice; with linseg, as low as .1
asig oscili kenv, ipitch, 2
out asig
endin
;SCO
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;copyright 1998 Paul M. Winkler, zarmzarm@hotmail.com
;****++++
;**** Last modified: Tue Aug 24 14:39:34 1999
;****----
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
f1 0 1024 2 1 0 ; impulse... works but sounds
bad
f2 0 32769 10 1 ; sine wave
t 0 80
i1 0 .25 100
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . >
i1 + . 50
s
t 0 100
i2 2 .1 100 30000
i2 3 . > .
i2 4 . > .
i2 5 . > .
i2 6 . > .
i2 7 . > .
i2 8 . > .
i2 9 . > .
i2 10 . > .
i2 11 . > .
i2 12 . 30 .
--
---------------- paul winkler ------------------
slinkP arts: music, sound, illustration, design, etc.
zarmzarm@hotmail.com --or-- slinkp AT ulster DOT net
http://www.ulster.net/~abigoo/
======================================================
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From: Torsten Anders
To: csound@maths.ex.ac.uk
Subject: Re: csound surround module: 8chan decoding
Date: Wed, 25 Aug 1999 00:11:36 +0200
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Hi,=20
although its really simple I needed a while to find the solution how to d=
ecode
the B-Format to eight channels. Here it is, if someone else may need it:
=20
a1 =3D gaw+gax+gay+gaz ; LEFT FRONT UPPER=20
a2 =3D gaw+gax-gay+gaz ; RIGHT FRONT UPPER
a3 =3D gaw-gax-gay+gaz ; RIGHT REAR UPPER
a4 =3D gaw-gax+gay+gaz ; LEFT REAR UPPER=20
a5 =3D gaw+gax+gay-gaz ; LEFT FRONT LOWER
a6 =3D gaw+gax-gay-gaz ; RIGHT FRONT LOWER
a7 =3D gaw-gax-gay-gaz ; RIGHT REAR LOWER
a8 =3D gaw-gax+gay-gaz ; LEFT REAR LOWER =20
outo a1, a2, a3, a4, a5, a6, a7, a8
gaw, gax etc. are the four channels of the B-Format, the numering depends=
on
our studio setup and may be changed of course.
Torsten Anders
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Message-ID: <37C31A37.99137632@intercom.es>
Date: Wed, 25 Aug 1999 00:18:31 +0200
From: Josep M Comajuncosas
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To: "17.hzV.tRL.478"
CC: csound@maths.ex.ac.uk
Subject: Re: detectbuggypittch.tch.tch.tchaschka.tchaj.
References: <199908241446.JAA08088@saba.wwa.com>
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Hmm why not, that=B4s a beautiful thinking
"17.hzV.tRL.478" escribi=F3:
> !mprov!ng odrz =3D 1eczusz !2 !mprov zelv +?
>
> > ... I=B4ll improve the tracker.
--
Josep M Comajuncosas
C/ Circumval.lacio 75 08790 Gelida - Penedes
Catalunya - SPAIN tel. 93 7792243
e-mail: gelida@intercom.es
ET Informatica de Sistemes
e-mail: jcomajuncosas@campus.uoc.es
http://members.tripod.com/csound/
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Reply-To: Rich Weisgerber
From: Rich Weisgerber
To: csound@maths.ex.ac.uk
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References: <99082500210803.00229@chewa>
Subject: Re: csound surround module: 8chan decoding
Date: Tue, 24 Aug 1999 17:19:03 -0700
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I'm in way over my head on this but. . .
page 321-322 of "Computer Music" (book) by Dodge & Jerse (ISBN =
0-02-864682-7) shows this instead.
I'm in the same boat you are, Torsten, and haven't figured this part =
out yet but here's what they say regarding B-format encode/decode.
Ambisonics B-format encode:
X =3D S * cos(aa) * cos(ea)
Y =3D S * sin(aa) * cos(ea)
Z =3D S * sin(ea)
W =3D 1/sqrt(2) * S
where:
S =3D input signal
aa =3D Azimuthal Angle
ea =3D Elevation Angle
Ambisonics B-format decode:
LFU =3D W + 1/sqrt(2) * (X + Y + Z)
RFU =3D W + 1/sqrt(2) * (X - Y + Z)
LBU =3D W + 1/sqrt(2) * (-X + Y + Z)
RBU =3D W + 1/sqrt(2) * (-X - Y + Z)
LFD =3D W + 1/sqrt(2) * (X + Y - Z)
RFD =3D W + 1/sqrt(2) * (X - Y - Z)
LBD =3D W + 1/sqrt(2) * (-X + Y - Z)
RBD =3D W + 1/sqrt(2) * (-X - Y - Z)
where:
L=3D Left
R =3D Right
F =3D Front
B =3D Back
U =3D Up
D =3D Down
I still don't have a very good handle on what W is, some kind of =
Omnidirectional characteristic, but Huh?
W isn't quite clicking in my head yet.. ;-)
X, Y, & Z are cartesian coordinates for the location of individual sound =
source, yes?
I imagine further discussion on whether this is accurate belongs in =
sursound list as they Are Ambisonics over there.
They'd probably be able to give us some clues on the licensing aspects =
of this information and how it's used foir what purpose.
However, I am also interested in getting this working in Csound for my =
own work, even at my inherent snail's pace.
We'll see how this goes... or not.. ;-)
r out
----- Original Message -----=20
From: Torsten Anders=20
To: csound@maths.ex.ac.uk=20
Sent: Tuesday, August 24, 1999 3:11 PM
Subject: Re: csound surround module: 8chan decoding
Hi,=20
although its really simple I needed a while to find the solution how =
to decode
the B-Format to eight channels. Here it is, if someone else may need =
it:
=20
a1 =3D gaw+gax+gay+gaz ; LEFT FRONT UPPER=20
a2 =3D gaw+gax-gay+gaz ; RIGHT FRONT UPPER
a3 =3D gaw-gax-gay+gaz ; RIGHT REAR UPPER
a4 =3D gaw-gax+gay+gaz ; LEFT REAR UPPER=20
a5 =3D gaw+gax+gay-gaz ; LEFT FRONT LOWER
a6 =3D gaw+gax-gay-gaz ; RIGHT FRONT LOWER
a7 =3D gaw-gax-gay-gaz ; RIGHT REAR LOWER
a8 =3D gaw-gax+gay-gaz ; LEFT REAR LOWER =20
outo a1, a2, a3, a4, a5, a6, a7, a8
gaw, gax etc. are the four channels of the B-Format, the numering =
depends on
our studio setup and may be changed of course.
Torsten Anders
------=_NextPart_000_0017_01BEEE54.C338DDA0
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charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
I'm in way over my head on this but. .=20
.
page 321-322 of "Computer Music" (book) =
by Dodge=20
& Jerse (ISBN 0-02-864682-7) shows this instead.
I'm in the same boat you are,=20
Torsten, and haven't figured this part out yet but here's =
what they=20
say regarding B-format encode/decode.
Ambisonics B-format =
encode: X =3D S *=20
cos(aa) * cos(ea)
Y =3D S * sin(aa) * =
cos(ea)
Z =3D S * sin(ea)
W =3D 1/sqrt(2) * S
where:
S =3D input signal
aa =3D Azimuthal Angle
ea =3D Elevation Angle
Ambisonics B-format =
decode:
LFU =3D W + 1/sqrt(2) * (X + Y + Z)
RFU =3D W + 1/sqrt(2) * (X - Y + Z)
LBU =3D W + 1/sqrt(2) * (-X + Y + Z)
RBU =3D W + 1/sqrt(2) * (-X - Y + Z)
LFD =3D W + 1/sqrt(2) * (X + Y - Z)
RFD =3D W + 1/sqrt(2) * (X - Y - Z)
LBD =3D W + 1/sqrt(2) * (-X + Y - Z)
RBD =3D W + 1/sqrt(2) * (-X - Y - Z)
where:
L=3D Left
R =3D Right
F =3D Front
B =3D Back
U =3D Up
D =3D Down
I still don't have a very good handle on what W is, some kind of=20
Omnidirectional characteristic, but Huh?
W isn't quite clicking in my head yet.. ;-)
X, Y, & Z are cartesian coordinates for the location of =
individual=20
sound source, yes?
I imagine further discussion on whether this is accurate belongs in =
sursound list as they Are Ambisonics over there.
They'd probably be able to give us some clues on the licensing =
aspects of=20
this information and how it's used foir what purpose.
However, I am also interested in getting this working in Csound for =
my own=20
work, even at my inherent snail's pace.
We'll see how this goes... or =
not.. =20
;-)
r out
----- Original Message -----
From:=20
Torsten Anders
To: csound@maths.ex.ac.uk
Sent: Tuesday, August 24, 1999 =
3:11=20
PM
Subject: Re: csound surround =
module:=20
8chan decoding
Hi,
although its really simple I needed a while =
to find=20
the solution how to decode
the B-Format to eight channels. Here it =
is, if=20
someone else may need it:
=20
a1 =20
=3D =20
gaw+gax+gay+gaz ; LEFT =
FRONT=20
UPPER
a2 =20
=3D =20
gaw+gax-gay+gaz ; =
RIGHT FRONT=20
UPPER
a3 =20
=3D =20
gaw-gax-gay+gaz ; =
RIGHT REAR=20
UPPER
a4 =20
=3D =20
gaw-gax+gay+gaz ; LEFT =
REAR=20
UPPER
a5 =20
=3D =20
gaw+gax+gay-gaz ; LEFT =
FRONT=20
LOWER
a6 =20
=3D =20
gaw+gax-gay-gaz ; =
RIGHT FRONT=20
LOWER
a7 =20
=3D =20
gaw-gax-gay-gaz ; =
RIGHT REAR=20
LOWER
a8 =20
=3D =20
gaw-gax+gay-gaz ; LEFT =
REAR=20
LOWER =20
=
=
;=20
outo a1, a2, a3, a4, a5, a6, =
a7,=20
a8
gaw, gax etc. are the four channels of the B-Format, the =
numering=20
depends on
our studio setup and may be changed of =
course.
Torsten=20
Anders
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Date: Tue, 24 Aug 1999 21:32:01 -0700
From: patrick pagano
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Subject: getting started with Granule
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Hi folks
been using Grain for a while and wanted to try to start to fool with
Granule for the pitch shift possibilities
but i keep getting an illegal combination between iskip and ilength
asig granule 3000, 64, 1, 0, 0, 3, 0, ?, .005, ?, 0.02, 50, 0.04, 50,
30, .31
can someone give me an appropriate settings or an idea of the
relationship?
thanks
Pat
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Message-ID: <000801beee9f$1e820200$4a99170c@default>
From: Hans Mikelson
To: Csound list
Subject: Re: Looking for drums
Date: Tue, 24 Aug 1999 21:11:17 -0500
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Hi,
Here is a drum machine. The kick isn't too bad, 808ish. Watch out for all
the line wraps. Sorry about the lack of comments.
Bye,
Hans Mikelson
;---------------------------------------------------------------------------
; Coded: February-May 1998 by Hans Mikelson
;---------------------------------------------------------------------------
; ORC
sr=44100
kr=22050
ksmps=2
nchnls=2
zakinit 50, 50
gareverb init 0
; Envelope
instr 5
idur = p3
iamp = p4
irate = p5
itab1 = p6
ioutch = p7
kenv1 oscili iamp, irate/idur, itab1
zkw kenv1, ioutch
endin
; Drum Machine
instr 30
idur = p3
iamp = p4
idurtab = p5
idrumtb = p6
iacctab = p7
iamptab = p8
ispeed = p9
isteps = p10
ikinch = p11
ioutch = p12
kstep init isteps-1
kspeed oscili 1, 1/p3, ispeed
loop1:
; Read table values.
kdur table frac((kstep + 1)/isteps)*isteps, idurtab
kdrnum table kstep, idrumtb
kaccnt table kstep, iacctab
kamp table kstep, iamptab
kaccnt1 table frac((kstep + 1)/isteps)*isteps, iacctab
kdur1 = kdur/kspeed ; Make the step smaller.
kdclick linseg 0, .002, 1, i(kdur1)-.004, 1, .002, 0
kaccin zkr ikinch
kaccnt = kaccnt*kaccin
kaccnt1 = kaccnt1*kaccin
; Switch between the different drums
if (kdrnum != 0) kgoto next1
; Noise Blip
kampenv0 linseg 0, .01/i(kaccnt1), iamp/2, .04/i(kaccnt1), 0, .01,
0
asig0 rand kampenv0
aflt1 butterlp asig0, 400*kaccnt
aflt2 butterbp asig0, 2000*kaccnt, 400*kaccnt
aout = aflt1*4+aflt2*2
kgoto endswitch
next1:
if (kdrnum != 1) kgoto next2
; Dumb Drum
kampenv1 expseg .0001, .01/i(kaccnt1), iamp, .08/i(kaccnt1), .01
arnd1 rand kampenv1
afilt reson arnd1, 500*kaccnt, 50*kaccnt
aout balance afilt, arnd1
kgoto endswitch
next2:
if (kdrnum != 2) kgoto next3
; Dumb Bass Drum
kampenv2 linseg 0, .02, iamp*2, .5/i(kaccnt1), 0
kfreqenv expseg 50, .05, 200*i(kaccnt1)+.001, .8, 10
arnd2 rand kampenv2
afilt1 butterbp arnd2, kfreqenv, kfreqenv/32
aosc1 oscil kampenv2, kfreqenv, 1
; afilt2 butterbp arnd2, kfreqenv*1.5, kfreqenv/32*1.5
; aosc2 oscil kampenv2, kfreqenv*1.5, 1
aout balance afilt+aosc1, arnd2*.5
kgoto endswitch
next3:
if (kdrnum != 3) kgoto next4
; KS Snare
kampenv3 linseg 0, .002, 1, i(kdur1)-.022, 1, .02, 0
kptchenv linseg 100, .01, 300, .2, 200, .01, 200
aplk3 pluck iamp/2, kptchenv, 50, 2, 4, 1/(1+i(kaccnt)), 3
aout = aplk3*kampenv3
kgoto endswitch
next4:
if (kdrnum != 4) kgoto next5
; FM Boink
kamp40 linseg 0, .002, 1, i(kdur1)-.004, 1, .002, 0
kamp41 linseg 0, .01, iamp, i(kdur1)-.06, iamp*.1, .05, 0
kamp42 linseg 0, .01, 1, i(kdur1)-.03, 0, .01, 0
kamp4 = kamp40*kamp41
klfo oscil 1, 8, 1
aout foscil kamp4, 110*kaccnt, 1, 1.5+klfo, kamp42*kaccnt, 1
kgoto endswitch
next5:
if (kdrnum != 5) kgoto next6
; Sorta Cool Knock Sweep Drum
kfreqenv51 expseg 800*i(kaccnt1)+.001, .04, 40, .01, 40
kfreqenv52 expseg 2000*i(kaccnt1)+.001, .04, 150, .01, 150
kampenv5 linseg 0, .001, iamp, .1, iamp*.5, .05, 0, .01, 0
asig rand kampenv5
afilt1 reson asig, kfreqenv51, kfreqenv51/32
afilt2 reson asig, kfreqenv52, kfreqenv52/16
aout1 balance afilt1, asig
aout2 balance afilt2, asig
aout = (aout1+aout2)/2
kgoto endswitch
next6:
if (kdrnum != 6) kgoto next7
; Ring Mod Drum 1
kampenv6 linseg 0, .01/i(kaccnt1), iamp, .04/i(kaccnt1), 0, .001, 0
kpchenv6 linseg 100, .01/i(kaccnt1), 400*i(kaccnt1), .04/i(kaccnt1),
100, .001, 100
asig1 oscil kampenv6, kpchenv6, 1
asig2 oscil kampenv6, kpchenv6*kaccnt, 1
aout balance asig1*asig2, asig1
kgoto endswitch
next7:
if (kdrnum != 7) kgoto next8
; Thunder Bomb
aop4 init 0
kae70 linseg 0, .002, iamp, i(kdur1)-.004, iamp, .002, 0
kae71 linseg 0, .1*i(kaccnt1), 1, .2*i(kdur1)-.1*i(kaccnt1), .8,
.8*i(kdur1), 0
kae72 linseg 0, .1*i(kaccnt1), 1, .2*i(kdur1)-.1*i(kaccnt1), .8,
.8*i(kdur1), 0
kae73 linseg 0, .1*i(kaccnt1), 1, .2*i(kdur1)-.1*i(kaccnt1), .8,
.8*i(kdur1), 0
kae74 linseg 0, .1*i(kaccnt1), 1, .2*i(kdur1)-.1*i(kaccnt1), .8,
.8*i(kdur1), 0
krnfqc randh 165*kaccnt1, 200
aop4 oscil 10*kae74, 3*(1+.8*aop4)*krnfqc, 1
aop3 oscil 20*kae73, .5*(1+aop4)*krnfqc, 1
aop2 oscil 16*kae72, 5.19*krnfqc, 1
aop1 oscil 2* kae71,.5*(1+aop2+aop3)*krnfqc, 1
aout tone aop1*kae70, 200*kaccnt1
next8:
if (kdrnum != 8) kgoto endswitch
; Sorta Cool Knock Sweep Drum 2
kfreqenv81 expseg 220*i(kaccnt1)+.001, .25/i(kaccnt1), 40, .01, 40
kampenv8 expseg .01, .001/i(kaccnt1), iamp, .002, iamp*.5, .06,
iamp, .05, iamp*.5, .1, .3*iamp, .1, .05*iamp, .01, .05*iamp
kdclick8 linseg 0, .002, 1, i(kdur1)-.04, 1, .002, 0
asig rand kampenv8
aflt reson asig, kfreqenv81, 1
abal balance aflt, asig
aout = abal*kdclick8
kgoto endswitch
endswitch:
; When the time runs out go to the next step of the sequence and
reinitialize the envelopes.
timout 0, i(kdur1), cont1
kstep = frac((kstep + 1)/isteps)*isteps
reinit loop1
cont1:
zaw aout*kdclick*kamp, ioutch ;, aout*kdclick*kamp
endin
;---------------------------------------------------------------------------
; Mixer Section
;---------------------------------------------------------------------------
instr 190
idur init p3
iamp init p4
inch init p5
ipan init p6
ifader init p7
asig1 zar inch ; Read input channel 1
kfader oscil 1, 1/idur, ifader
kpanner oscil 1, 1/idur, ipan
kgl1 = kfader*sqrt(kpanner) ; Left gain
kgr1 = kfader*sqrt(1-kpanner) ; Right gain
kdclick linseg 0, .002, iamp, p3-.004, iamp, .002, 0 ; Declick
asigl = asig1*kgl1 ; Scale and sum
asigr = asig1*kgr1
outs kdclick*asigl, kdclick*asigr ; Output stereo pair
endin
;---------------------------------------------------------------------------
; Zak Clear
;---------------------------------------------------------------------------
instr 195
zacl 0, 50
zkcl 0, 50
endin
;SCO
f1 0 8193 10 1 ; Sine
f2 0 1025 7 1 1025 1 ; 1?
f3 0 1025 -7 5 1025 5 ; Speed
f4 0 1025 10 1 0 .333 0 .2 0 .143 0 .111 0 .0909 ; Square ?
f90 0 1025 -7 0 1025 1 ; Ramp, FadeIn, PanRL
f91 0 1025 -7 1 1025 0 ; Saw, FadeOut, PanLR
f92 0 1025 -7 1 1025 1 ; Const 1, PanL
f93 0 1025 -7 .5 1025 .5 ; Const .5, PanC
f94 0 1025 -7 0 1025 0 ; Const 0, PanR
f95 0 1025 -7 .8 513 .2 512 .8 ; PanRLR
f96 0 1025 -7 .2 513 .8 512 .2 ; PanLRL
f97 0 1025 -7 1 1025 .7 ; FadeDown
f98 0 1025 -7 .5 1025 1 ; FadeUp
f99 0 1025 -7 0 513 1 512 1 ; FadeIn & Hold
;---------------------------------------------------------------------------
; Drums
;---------------------------------------------------------------------------
; Envelope
f10 0 1025 -7 1.2 128 1.5 128 1.7 256 1 513 1.3 ; Accent Envelope
; Sta Dur Amp Rate Table OutKCh
i5 0 25.6 1 1 10 1
; Drums : 0=HiHat, 1=Tap, 2=Bass, 3=KS Snare, 4=FMBoink, 5=Sweep,
6=RingMod1
; Beat 1 2 3 4 5 6 7 8 9 10
11 12 13 14 15 16 17
f21 0 32 -2 2 2 2 2 2 2 2 2 2 2 2
2 2 2 2 1 1 ; Duration
f22 0 32 -2 8 8 8 8 8 8 8 8 8 8 8
8 8 8 8 8 8 ; Drum
f23 0 32 -2 1.5 1.1 1.1 .98 .99 1.1 1 1.5 .9 .8 1
1 1 1 1 .8 .8 ; Accent
f24 0 32 -2 1.2 .8 1 1 1 1 1 1.2 1.3 .5 1
1 1 1 1 1 1 ; Amp
f25 0 32 -2 1.2 1 .8 1 1 1 1 1.2 .5 1.3 1
1 1 1 1 1 1 ; Amp
; Sta Dur Amp DurTab DrumTab Accent AmpTab Speed Steps InKCh
OutCh
i30 0 25.6 8000 21 22 23 24 3 17 1 1
i30 0 25.6 8000 21 22 23 25 3 17 1 2
; Mixer
; Sta Dur Amp Ch Pan Fader
i190 0 25.6 2.0 1 92 92
i190 0 25.6 2.0 2 94 92
; Clear Zak
; Sta Dur
i195 0 25.6
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Date: Wed, 25 Aug 1999 05:55:57 +0200
From: rasmus ekman
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To: Csound list
Subject: Re: getting started with Granule
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patrick pagano wrote:
>
> been using Grain for a while and wanted to try to start to fool with
> Granule for the pitch shift possibilities
> but i keep getting an illegal combination between iskip and ilength
Hm... The docs seem pretty clear here: igskip and ilength says
what section of the (mono soundfile) wavetable you want to use.
igskip = seconds into the table, ilength = seconds of table to
use, starting from igskip. So the table must at least be of
the length igskip+ilength. Now I haven't tried it in quite
a while, but I presume those "seconds" are relative to sr - if
sr=44100, one second should be 44100 sample points in the table.
I've always preferred fof2/fog for granular synthesis, while some
recommend sndwarp. These ugens give k-rate control over the starting
point of grains (allowing both time-stretching / pitch shifting
and randomised granular textures), while granule has most interesting
params as i-time variables. It might faster for dense textures though.
But not for me to tell you how to make music; just have fun.
rrrasarsuremmmrsrmurese
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To: csound@maths.ex.ac.uk
From: Arne Hanna
Subject: Kpow and Apow
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Hi Folks. What happened to kpow and apow? These opcodes are listed in the
manual but any attempt at using either of them is greeted with an unknown
opcode message. I'm using the Mills PPC version of Csound; Its not the
most recent version, but 3.5 something? I think. In any case, these two
opcodes have been around for a while now; one of Andr=E9 Bartetski's Cmask
examples uses kpow.
Anyone?
Cheers
Arne
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From: Josep M Comajuncosas
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CC: csound@maths.ex.ac.uk
Subject: Re: Insisting on PITCH
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"Job M. van Zuijlen" escribi=F3:
> I have had reasonable success with the pitch tracking using LPC
> analysis.
Yes it works fine with quite stable signals and a good initial estimation=
. It
would be useful to have its pitchtracker as an independent orchestra
opcode...
> Not to
> disillusion you, but don't feel too bad if you haven't solved this
> problem in one afternoon.
I realized that already... better to spend the time in my instruments. Bu=
t if
only I had a working pitch tracker... ... ...
--
Josep M Comajuncosas
C/ Circumval.lacio 75 08790 Gelida - Penedes
Catalunya - SPAIN tel. 93 7792243
e-mail: gelida@intercom.es
ET Informatica de Sistemes
e-mail: jcomajuncosas@campus.uoc.es
http://members.tripod.com/csound/
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From: Peter Neubacker
Date: Wed, 25 Aug 99 10:12:47 +0200
To: csound@maths.ex.ac.uk
Subject: Re: new adsynt opcode
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I put a version of DirectCsound compiled with my
new opcodes adsynt and hsboscil for download to:
http://www.harmonik.de/femc/femcdown.html
With DirectCsound, I get about 2000 individual
sines running with adsynt in realtime...
(Gabriel, could you build them into your release?)
I am preparing an example orchestra!
Peter
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From: Peter Neubacker
Date: Wed, 25 Aug 99 10:15:13 +0200
To: csound@maths.ex.ac.uk
Subject: Re: new adsynt opcode
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>With DirectCsound, I get about 2000 individual
>sines running with adsynt in realtime...
P.S.: on a Pentium II 400
Peter
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From: Peter Neubacker
Date: Wed, 25 Aug 99 10:26:01 +0200
To: csound@maths.ex.ac.uk
Subject: schedule
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schedule does work!
The hint from Gerard van Dongen
to reinit it was the point!
Below you find an example
of an auto-composing orc.
Peter
;====================================================================
; AUTORUN.SCO
;--------------------------------------------------------------------
f 1 0 1024 -10 1 0.5 0.1 0.2 0.3
f 0 3600 ; run for ... seconds
i 1 0 1 ; start realtime composing instrument
e
;====================================================================
; AUTORUN.ORC
;--------------------------------------------------------------------
sr = 22050
kr = 220.5
ksmps = 100
nchnls = 2
giwave = 1
;====================================================================
instr 1 ; example event generating instrument: a simple fractal loop
;--------------------------------------------------------------------
ihold ; Run forever
;
kx1 init 0 ;
kx2 init 0 ;
;
idur = 1.5 ;
ispeed = 6 ;
;
ktime phasor ispeed ; Timer for
ktrig trigger ktime, 0.5, 0 ; event generation
if (ktrig<0.5) goto noevent ;
;
kx1 = kx1 * kx1 - 1.74 ; Some simple fractal
kx2 = kx2 * kx2 - 1.7499 ; parameter generation
kamp = 0.05 + (kx2+2) / 4 * 0.9 ;
kcps = 200 + 100 * kx1 / 2 ;
;
igoto noevent ; Don't schedule
initsched: ; at first init
schedule 2, 0, idur, i(kamp), i(kcps) ;
reinit initsched ; reinit schedule
; for each event
noevent: ;
;
endin ;
;====================================================================
instr 2 ; sound generating instrument
;--------------------------------------------------------------------
kenv expon 1, 1, 0.01
kamp = 20000*p4*kenv
asig oscili kamp, p5, giwave
outs asig, asig
endin
;--------------------------------------------------------------------
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From: Peter =?iso-8859-1?Q?Neub=E4cker?=
Date: Wed, 25 Aug 99 10:40:32 +0200
To: csound@maths.ex.ac.uk
Subject: Re: new adsynt opcode
Reply-To: peter@orpheus.selene.cube.net
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> I put a version of DirectCsound compiled with my
> new opcodes adsynt and hsboscil for download to:
> http://www.harmonik.de/femc/femcdown.html
Sorry, due to some changes it will be available tomorrow!
Peter
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From: rasmus ekman
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Subject: Re: Kpow and Apow
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Arne Hanna wrote:
>
> Hi Folks. What happened to kpow and apow? These opcodes are listed in the
> manual but any attempt at using either of them is greeted with an unknown
> opcode message.
They were renamed to pow.
Get the new manual:
http://www.geocities.com/~csoundmanual
re |