| Gabriel Maldonado wrote:
>I think that writing an a-rate freq-cutoff version of csound filter should
be a quite
>trivial job. Maybe I will do this in the future, because I'm curious about
the sounds that
>can be produced. Are you sure that they are interesting?
I think you can get pretty close to this by using a k-rate at 22050 and then
running a k-rate oscillator at audio frequencies to modulate Fco.
Trying to implement this I came up with a question:
The test for a-rate is done on XINCODE. For example grain uses the
following:
p->ampadv = (p->XINCODE & 0x2) ? 1 : 0;
p->lfradv = (p->XINCODE & 0x1) ? 1 : 0;
p->dnsadv = (p->XINCODE & 0x4) ? 1 : 0;
The opcode is called with:
aout grain xamp, xfrqc, xdns, ...
I would have thought the test for amplitude would have been on the first bit
1, frqc on 2, density on 4, the next one on 8, etc. Can someone explain how
XINCODE works?
Thanks,
Hans Mikelson
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From: Sean Costello
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To: Gabriel Maldonado
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Csound List
Subject: Re: a-rate modulating filters (was Re: CSound book)
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Gabriel Maldonado wrote:
>
> I think that writing an a-rate freq-cutoff version of csound filter should be a quite
> trivial job. Maybe I will do this in the future, because I'm curious about the sounds that
> can be produced. Are you sure that they are interesting?
Well, I do it in the analog realm all the time (the Multimoog, Octave
Cat, and Prophet 5 are a few of the synths that feature this ability).
It is a very nice feature. With lower amounts of resonance and a higher
cutoff frequency, the output resembles a "dirtier" version of the
input. With a lower cutoff, nice ring-modulation effects can occur.
With higher resonance, the sound is close to traditional FM. The nice
thing about audio rate filter modulation is that any sound can be used
for the input, and the output will retain the original pitch of the
input sound (at least with filters that are not in self-oscillation).
It works very well on drum sounds.
Sean Costello
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Date: Mon, 28 Dec 1998 19:19:06 +0200
From: Yair Kass
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Hi list,
This is something i encountered when moving
from Direct2.1 to Direct2.11.
I wouldn't say it's a bug (maybe an improvement)
but i'm reporting this for those who may be interested:
The code below plays a sine wave while the user
controls the depth of a vibrato.
Direct2.1 didn't mind that 'kenv' is used
(as the max. value for 'kenvdp') before it is defined.
But Direct2.11 DID.
To fix that, just put the 'kenvdp ctrl7.........' line
AFTER 'kenv' is defined.
I know this may sound (or be) very trivial but since
there was a difference between the two version's
performance, I thought I'd say something.
Hope this helps (:-}
Yair
;------------------------------------------------------------
sr = 44100
kr = 441
ksmps = 100
nchnls = 1
instr 1
ihold ;Keep instrument on.
kenvdp ctrl7 1,1, 1,kenv ;Controls the depth of the vibrato
kenv oscili 0.5,2,1 ;Vib -
kenv = kenv+0.5 ; - rato
asig oscili 20000,440,1 ;Carrier
aout = asig*kenvdp ;Modulate
out aout
endin
;--------------------------------------------------------------
f1 0 4096 10 1 ;Simple sine wave
f0 600
i1 0 1
e
;--------------------------------------------------------------
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From: Paul Winkler
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Kevin Gallagher wrote:
>
> Thanks to everyone who has replied to my query about midi networking. I
> will try this and report how it turns out!
>
> I have another question, and this actually pertains more or less to
> csound.:
>
> I pose this to PC users: when I write a script that pans hard left or
> hard right, the result is a sound that is not fully in the specified
> speaker. It's more like a 70/30 split. It's not just csound either...I
> tried a bunch of applications. Each time I make sure sound is being sent
> to one side and nothing is being sent to the opposite side, I get the
> 70/30 split rather than 100/0. Some machines, however, work okay in that
> I CAN get hard left or right. I called IBM tech support and they were
> not too helpful. Has anyone here experience this problem with stereo
> imaging? I have a PnP Audio Crystal FM Synthesis soundcard (built into
> the system) and I wonder if it's my card that's causing the problem.
One possibility is that your soundcard has some sort of "surround sound"
going on there. If so, there should be a way to turn it off. With my
soundcard (a Turtle Beach Malibu, which is also based on a Crystal
chip), I've noticed that if the "SRS 3-D Surround Sound" is turned on,
panning does weird and unwanted things much like you describe. It also
adds a lot of hiss, so I always have it turned off.
--PW
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Date: Tue, 29 Dec 1998 10:22:08 +0100
From: Lemoine Jean-Pierre
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The site is temporary done, as the servers are being moved from one
location to another. This site is hosting by a friend acting like an isp
on his spare time, thus this operation may take some time. Anyway, I
hope everyting will be up soon, and with an upgraded version of
HPKComposer I am just finishing: usability enhancement, 3.49 opcodes
support.
Seasons Greetings
jean-pierre
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Date: Tue, 29 Dec 1998 00:41:04 +0100
From: Josep M Comajuncosas
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Subject: Re: Pink Noise?
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Hi,
there was a short talk about this topic at music.dsp some weeks ago. Here
you have two -3dB/oct filter (approximations!) posted to that list. You can
port them to csound with rand, biquad and kfilter2, or coding them directly
of course.
Hope this helps,
Josep M Comajuncosas
How to make pink noise
----------------------
This is an approximation to a -3dB/oct filter using a weighted sum
of 6 first order low-pass filters. The weighting coefficients (the
second number on each line) need to be scaled to suit your
application (ie. the amplitude of your white noise input and what
peak or rms output level you want).
...
white = (double)(rand() - HALF_RAND_MAX);
buf0 = 0.997 * buf0 + 0.029591 * white;
buf1 = 0.985 * buf1 + 0.032534 * white;
buf2 = 0.950 * buf2 + 0.048056 * white;
buf3 = 0.850 * buf3 + 0.090579 * white;
buf4 = 0.620 * buf4 + 0.108990 * white;
buf5 = 0.250 * buf5 + 0.255784 * white;
pink = buf0 + buf1 + buf2 + buf3 + buf4 + buf5;
an equiripple approximation to the ideal pinking filter can be realized by
alternating real poles with real zeros. a simple 3rd order solution that i
obtained is:
pole zero
---- ----
0.99572754 0.98443604
0.94790649 0.83392334
0.53567505 0.07568359
the response follows the ideal -3 dB/octave curve to within + or - 0.3 dB
over a 10 octave range from 0.0009*nyquist to 0.9*nyquist. probably if i
were to do it over again, i'd make it 5 poles and 4 zeros.
> At 02:58 PM 12/27/98 -0500, Brandon Nelson wrote:
> >Can anyone tell me how to create pink noise? I've designed an instrument
> >that uses Butterworth filtered noise to create a pipe-like sound, and it
> >took me forever to realize why the low notes had no energy.
|