| Hi,
thought it would be a good idea to use the precise oscillator opcode poscil of
FM, but in fact it is really not, the opcode produces awful distortion or
similar treated in that way. poscil3 is only a little better. Am I missusing it
or is that a bug?
sr = 44100
kr = 44100
ksmps = 1
instr 3 ; works well with old oscili
idur = p3
iamp = 20000
ifqc = cpspch(8.00)
icar = 1
imod = 2.7
indx = 4
ifn = 1
kmod oscili indx*imod*ifqc, ifqc*imod, ifn ; modulator
kenv linen iamp, .1, idur, .1
a1 oscili kenv, ifqc*icar+kmod, ifn ; carrier
out a1
endin
instr 4 ; sounds really bad
idur = p3
iamp = 20000
ifqc = cpspch(8.00)
icar = 1
imod = 2.7
indx = 4
ifn = 1
kmod poscil indx*imod*ifqc, ifqc*imod, ifn ; modulator
kenv linen iamp, .1, idur, .1
a1 poscil kenv, ifqc*icar+kmod, ifn ; carrier
out a1
endin
BTW: oscil3 seems to interpolate to well, it reduces the signal almost to
sine waves...
instr 5
idur = p3
iamp = 20000
ifqc = cpspch(8.00)
icar = 1
imod = 2.7
indx = 4
ifn = 1
kmod oscil3 indx*imod*ifqc, ifqc*imod, ifn ; modulator
kenv linen iamp, .1, idur, .1
a1 oscil3 kenv, ifqc*icar+kmod, ifn ; carrier
out a1
endin
Torsten Anders
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Date: Wed, 25 Aug 1999 09:01:01 -0700 (PDT)
From: "Matt J. Ingalls"
To: Rich Weisgerber
cc: csound@maths.ex.ac.uk
Subject: Re: csound surround module: 8chan decoding
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im probably not the best to respond to all this, but oh well:
> page 321-322 of "Computer Music" (book) by Dodge & Jerse (ISBN 0-02-864682-7) shows this instead.
wow. im surprised its in there!
>
> Ambisonics B-format encode:
i might point out that this encode formula is for panning a mono source,
which i am guessing would sound pretty much like a
chowning-type "mover" orchestra...(although you have the speakers
working together rather than just 1 or2)
there are methods to take a mono source and synthesize image "width" - but
maybe someone else would be better to explain this...
>
> I still don't have a very good handle on what W is, some kind of
>Omnidirectional characteristic, but Huh?
yes - one way to think of it is as 4 microphones stacked on top of each
other: 3 figure of 8s pointed forward, left, and up and one
omni. (basically thats what i soundfield mic does - although i think the
capsules are arranged in a tetragon[or whatever the 4 sided regular
polygon-thing is called] and then does a translation to B-Format)
>
> I imagine further discussion on whether this is accurate belongs in
>sursound list as they Are Ambisonics over there.
it doesnt really occur that much on sursound list (but it should!) - most
of sursound list talk is about recording/etc.. not
synthesis/manipulation/etc
> 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.
as far as i understand, B-format/etc patent has run out and is public
domain - the only current patents exist for converting B-Format into Dolby
Surrond (g-Format)
>
-matt
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Date: Wed, 25 Aug 1999 13:35:39 EDT
Subject: Fwd: getting started with Granule
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Pat-
If I could see what you have been using for igskip and ilength parameters
that would help troubleshooting, but here's some thoughts:
your source soundfile must be longer than ilength + igskip, be sure to take
igskip_os into account, this could put you over.
your igskip_os (*offset*) is .005 (sec), if igskip is 0 this might cause
problems ...?
if that doesn't work, ask again, post the full instrument and the ftable (not
necessarily the soundfile, but the ftable values) and I'll look more.
also, 64 voices may take a long time to compute (if it doesn't now, when you
get longer files or more instruments it will). I've found that using 15 or
20, or even 10, voices sounds about the same and runs a lot faster. Even if
you decide you want a lot of voices in the end, it can make tweaking the
instrument a lot easier/faster.
Good luck-
nate
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 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 >>
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Date: Tue, 24 Aug 1999 21:32:01 -0700
From: patrick pagano
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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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Date: Wed, 25 Aug 1999 19:35:51 +0200
From: Gabriel Maldonado
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Subject: Re: new adsynt opcode
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I will put your great opcode in next DirectCsound version,
congratulations!
P.S. How do you intend to continously vary parameters stored in the
table?
Gab
Peter Neubacker wrote:
> 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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Date: Wed, 25 Aug 1999 17:50:58 -0700 (PDT)
From: Wayne Freno
To: Csound List
Subject: Scores on paper
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I have heard about the importance of having a printed score
if one is going to get one's music either heard or performed
(even tape music I understand) and I am curious about how
people on the list go about this. Do any of you use a graphic
representation, (or traditional notation etc), of your csound
scores on paper?
Wayne
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From: Larry Troxler
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Subject: wgpluck : unwanted freq quantization?
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Hi all, it seems that either the wgpluck opcode is not generating the
compensating all-pass filter correctly (I assume that this opcode is a
translation of Perry Cook's pluck instrument), or I am doing some thing
more basically wrong. See the PLUCKM instrument in the following orc,
and let me know if there's anything wrong.
I do think it's some sort of frequency quantization problem, because it
gets worse as the pitch gets higher.
There are more than one opcodes, it seems, for a physically modeled
pluck, so if wgpluck is not the one I should use, let me know, please!
Thanks :-) Really, I'm looking for the equivalent of Cook's pluck
instrument that he included with his C++ toolkit - it generates an
all-pass filter in the feedback loop which implements a
fractional-sample delay, to correct the frequency quantisation which
would otherwise result from using a delay line which is fixed at an
integer number of samples.
Thanks in advance, people!!
~~~~
ksmps = 10
nchnls= 1
#define PLUCKM #1#
#define PLUCK1 #2#
instr $PLUCK1
aexcite init 0.0
a1 wgpluck p4, 0.4, p5, p6, p7, p8, aexcite
a2 = 32000 * a1
out a2
endin
instr $PLUCKM
aexcite init 0.0
inum notnum
icps cpsmidi
a1 wgpluck icps, 0.3, 0.35, 0.12, 5, 20, aexcite
out 32000 * a1
endin
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Message-ID: <000701beef7d$c6897440$2799170c@default>
From: Hans Mikelson
To: Csound list
Subject: Another drum
Date: Wed, 25 Aug 1999 23:45:07 -0500
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Hi,
Here is another drum. This one is much simpler and should be easier to
modify and play around with, and perhaps sounds fatter. It basically takes
noise, sweeps a resonant low-pass filter over it, sweeping both frequency
and resonance with exponential envelopes, applies an amplitude envelope and
finally applies compression/limiting. The p-fields should be useful for
generating a variety of sounds. Try changing the limiting curve for more or
less compression.
Happy Csounding,
Hans Mikelson
; ORC
;---------------------------------------------------------
; Kick Drum
; Coded by Hans Mikelson August 25, 1999
;---------------------------------------------------------
sr = 44100 ; Sample rate
kr = 44100 ; Kontrol rate
ksmps = 1 ; Samples/Kontrol period
nchnls = 2 ; Normal stereo
zakinit 50,50 ; May need this later
;---------------------------------------------------------
; Kick Drum 1
;---------------------------------------------------------
instr 10
idur = p3 ; Duration
iamp = p4 ; Amplitude
iacc = p5 ; Accent
irez = p6 ; Resonance
kfenv expseg 1500*iacc, .04, 200, .04, 120, idur-.04, 40 ; Freq
Envelope
krenv expseg .7, idur, 1 ; Q Envelope
kaenv expseg .1, .001, 1, .02, 1, .04, .7, idur-.062, .7 ; Amp Envelope
kdclck linseg 0, .002, 1, idur-.004, 1, .002, 0 ; Declick
asig rand 2 ; Random
number
; I use rand 2 to rely on moogvcf built in limiter
aflt moogvcf asig, kfenv, irez*krenv ; Moog filter
aout1 = aflt*kaenv*3 ; Scale the sound
krms rms aout1, 1000 ; Limiter, get rms
klim table3 krms*.5, 5, 1 ; Get limiting value
aout = aout1*klim*iamp*kdclck ; Scale again and ouput
outs aout, aout ; Output the sound
endin
; SCO
f1 0 65536 10 1
f5 0 1024 -8 1 256 1 256 .7 256 .1 256 .01 ; Compression/Limiting curve
; Sta Dur Amp Accent Q
i10 0.0 .18 24000 1 1.2
i10 0.5 . . .8 1.5
i10 1.0 . . .5 1.8
i10 1.5 . . .4 2.2
i10 2.0 .18 24000 1 1.2
i10 2.5 . . .8 1.5
i10 3.0 . . .5 1.8
i10 3.75 . . .4 2.2
i10 3.5 . . .4 2.2
i10 4.0 .18 24000 1 1.2
i10 4.5 . . .8 1.5
i10 5.0 . . .5 1.8
i10 5.5 . . .4 2.2
i10 6.0 .18 24000 1 1.2
i10 6.5 . . .8 1.5
i10 7.0 . . .5 1.8
i10 7.75 . . .4 2.2
i10 7.5 . . .4 2.2
|