| >sorry for "loop" but I'd like to ask it again:
>is it possible to know characteristics (cutoff frequency,
>e.t.c.) of the LPF:
>
>a3 delay1 a2
>afeedback=(a2+a3)*.5 ?
>
as always, I should let people who really know about these things speak.
but sergey hasn't had an answer on this, and it's a simple one, so I'll try a
basic answer.
the output the simple averaging lowpass filter of the type
y(n) = 0.5 * x(n) + 0.5 * x(n-1)
has an amplitude of 1 at 0Hz, and decays constantly to amplitude 0 at sr/2,
that is, the Nyquist frequency.
the response curve is the first cuadrant of a cosine.
the cutoff frequency (the -3dB point) is sr/4.
conversely, with
y(n) = 0.5 * x(n) - 0.5 * x(n-1)
you have a highpass filter of inverse response curve: amplitude 0 at 0Hz,
amplitude 1 at sr/2, Fc at sr/4.
these are of course particular cases of non-recursive filters, you should try
to read something about FIRs and IIRs, to get more complete picture of digital
filtering.
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From: jpff@maths.bath.ac.uk
Subject: Re: Bug-report and fix (was: Frequency Shifting & Hilbert Transforms)
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Message written at 19 Mar 1999 09:44:50 +0000
i have added that fix to both diskin and soundin. Thank you to both
Richard Dobson and matt ingalls.
==John ffitch
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Date: Fri, 19 Mar 1999 16:07:57 +0100
From: Roger Klaveness
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Subject: Re: Frequency Shifting & Hilbert Transforms
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the error is mine i'm afraid :-| ( blush blush )
setting kr=sr should be correct though
Roger
Jens Groh wrote:
>
> > > > axv0 = axv1
> > > > axv1 = axv2
> > > > axv2 = axv3
> > > > axv3 = axv4
> > > > axv4 = axv5
> > > > axv5 = axv6
> > > > axv6 = axv7
> > > > axv7 = axv8
> > > > axv8 = axv9
> > > > axv9 = axv10
> > > > axv10 = ain/igain;
> > >
> > > [...] replace your delay line with one that has its taps at every sample,
> > > not every ksmps samples. Try it with the delay1 opcode.
> >
> > ??
> > I'm using a-rate for the filter
> >
>
> Roger, I'm afraid there is a big misunderstanding! (Hope it's not on my side.)
> The a-rate operations in Csound code are _not_ executed every a-rate cycle.
> They are executed every k-rate cycle, but apply to a _vector_ of values.
> (The dimension of this vector is ksmps.)
>
> Jens
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Date: Fri, 19 Mar 1999 09:51:55 -0800 (PST)
From: "Matt J. Ingalls"
To: pete moss
cc: csound
Subject: Re: csounders at seamus99
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im not officially going - but was thinking of driving down on saturday to
hook up with people. say hi. i will most likely look like this:
http://www.concentric.net/~Mingalls/photos/door.jpg
-matt
On Thu, 18 Mar 1999, pete moss wrote:
> are anyone of you csounders out there going to be at seamus99 next week
> in san jose? i will be there. i have a piece called 'descent' on the
> first concert at 1:30 on friday. look for me. it would be nice to meet
> some of my fellow csound users face to face.
>
> pete
>
>
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Date: Fri, 19 Mar 1999 09:55:50 -0800
From: Erik Spjut
Subject: Re: simple LPF
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Frequency response is:
(1 + exp(-j omega))/2
which has a magnitude response of:
Cos(Pi f/sr)
or 1 at f/sr=0, -3dB at f/sr=0.25, 0 at f/sr=0.5
And a phase response of
phi in degrees = -180 f/sr
f is frequency, sr is sampling rate. Anything else?
At 1:15 PM +0300 3/19/99, Sergey Batov wrote:
>Hi,
>sorry for "loop" but I'd like to ask it again:
>is it possible to know characteristics (cutoff frequency,
>e.t.c.) of the LPF:
>
>a3 delay1 a2
>afeedback=(a2+a3)*.5 ?
>
>Thank you,
>Sergey Batov batov@glasnet.ru
Dr. R. Erik Spjut
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From: Hans Mikelson
To: csound@maths.ex.ac.uk
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Subject: Re: Okay, now your fav' MATH books survey!
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Hi,
My favorite recreational mathematics books are by Clifford Pickover. He has
many mind expanding (shattering) ideas. These may not be that good for
orderly learning of mathematical concepts but are good for stimulating the
imagination.
Regards,
Hans Mikelson
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From: Hans Mikelson
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Subject: Re: pitch in K-S algorithm
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Hi,
Rewriting the second example yields the following instrument which still has
the tuning problem:
instr 2
icps = cpspch(p4)
iamp = p5
kamp linseg iamp, 1/icps, iamp, 0, 0, p3-1/icps, 0
asig init 0
anoise rand kamp
adel delayr 1/icps
delayw asig
asig tone adel+anoise,sr/2
outs ganull, asig
endin
Making the following change solves the tuning problem.
adel delayr 1/icps
asig tone adel+anoise,sr/2
delayw asig
Setting sr=kr also solves the problem.
Hope this helps,
Hans Mikelson
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Date: Fri, 19 Mar 1999 23:37:15 -0600
From: pete moss
Organization: pete moss GmbH
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Subject: Korrepetition
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any germans on this list? what does Korrepetition mean in english?
sorry for wasting bandwidth, but none of my local german friends know
any musical terms. email me privately.
thanks
pete
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From: Sergey Batov
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Subject: thanks
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Thank you all for help in my fight against ignorance :-).
Regards,
Sergey Batov batov@glasnet.ru
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From: Jean-Michel DARRMONT
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Subject: Re: parameters in K-S?
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It's possible with wgpluck2. You can actually vary place of pluck and place of
hearing at k-rate.
Sergey Batov wrote:
>
> Dear Csounders!
> I'd like to know: is it possible to change "place of plucking"
> and time of decay in a simplest form of Karplus-Strong algorithm
> ("cycling" delay line + LPF)?
--
Jean-Michel DARREMONT
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From: Jean-Michel DARRMONT
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Hi CSounders,
wgpluck2 sounds really great!
Just a small disapointment:
pitch is not a k argument.
How is it possible to vary pitch at krate for making vibratos, bends, slurs etc....
It would bring live to it.
Thanks for any avice.
--
Jean-Michel DARREMONT
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Subject: Cage's Williams Mix
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Hi,
Reading the book:"Conversing with Cage" from Richard Kostelanetz I notice=
d
Cage's commentary about his
realisation of Williams mix in 1952.
They choped up a recorded tape in 1097 fragments and spliced them back in=
to
the band.
In that way they put the splices in any orientation refered to the normal
horizontal reading.
The splices where played mainly diagonaly.
He said that the sounds produced that way were "perfectly beautiful sound=
s" no
doubt they are at least quite unusual.
Here comes to mind this question: how can a soundfile be read in CSound a=
t a
variable angle saying that 0=B0 is the normal playback, 180=B0 backward a=
nd 360=B0
normal playback again?
It would be interesting to try this, specialy when we consider they spent=
one
year with a five or six persons team to realize Williams Mix, cutting,
splicing tiny pieces of tape, using chance operations to determine length=
and
angle of reading in a terribly meticulous work.=20
Digital synthesis could do that in a clic and that way experience and bri=
ng
the process further.
Is hetro/adsyn necessary or pvoc or something simpler?
Any idea?
Regards.
--=20
Jean-Michel DARREMONT
|