| Message written at 06 Apr 1998 23:27:08 +0100
In-reply-to: <199804041938.LAA17411@eos.arc.nasa.gov> (jims@eos.arc.nasa.gov)
I originlly wrote teh heartbeat function to use . characters, but i
was prevailed upon to produce the rotating bar. Is there a consensus,
or do you want this as a controlled option? Advice please.
==John ffitch
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Date: Tue, 07 Apr 1998 14:43:10 +0200
From: rasmus ekman
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To: Csound list
Subject: Re: Winsound realtime
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DJTA Quinn wrote:
>
> Can winsound be used like csound at the command line for realtime
> (or faketime for that matter)?
Yes. Or you can set the corresponding options in the GUI.
> ....Actually, can the GUI Winsound do realtime?
Try it. If it doesn't work, try Maldonado's version (I don't know to
what extent his work has been integrated into the Bath distribution).
http://www.agora.stm.it/G.Maldonado/download.htm
hth,
re
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From: John Michael Davison
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Subject: Re: humor, PC police, & SG album Csound joke thread :)
To: csound@maths.ex.ac.uk
Date: Tue, 7 Apr 1998 08:46:20 -0400 (EDT)
Cc: jim-ra@eos.arc.nasa.gov
In-Reply-To: <199804062051.NAA08933@eos.arc.nasa.gov> from "Jim Stevenson's reader" at Apr 6, 98 01:51:23 pm
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Jim Stevenson ("jim-ra@eos.arc.nasa.gov") writes:
> >From ingalls@ella.mills.edu Sat Apr 4 11:23:54 1998
> id LAA17056 for ; Sat, 4 Apr 1998 11:23:51
> -0800
[rest of junk deleted]
I can see why Barry Vercoe killed the original Csound mailing list.
There's enough silly, irrelevant garbage on this list to render it almost
unreadable.
--
John Davison
davisonj@panix.com
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From: Jon Christopher Nelson
Date: Tue, 7 Apr 98 08:36:05 -0600
To: Contribute
Subject: still some slots available . . .
References: <199804051228.RAA14486@bom2.vsnl.net.in>
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For anyone who would like to study Csound, MAX, KYMA, or
algorithmic composition amidst the scenic beauty of the rocky
mountains, there are still a few slots available in the CEMI (Center
for Experimental Music and Intermedia) Digital Soundscapes Summer
Workshops. If interested, check out our web site at
http://www.music.unt.edu/CEMI/cb or contact Jon Nelson at
jnelson@sndart.cemi.unt.edu
Jon Christopher Nelson, Director
CEMI: Center for Experimental Music and Intermedia
University of North Texas College of Music
Denton, TX 76203
ph. 940-369-7531
fax 940-565-2002
jnelson@sndart.cemi.unt.edu
http://www.music.unt.edu/comp/jnelson.htm
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From: jpff@maths.bath.ac.uk
To: coble@panix.com
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In-Reply-To: <3528D0AC.36F59FD2@panix.com> (message from Michael Coble on Mon,
06 Apr 1998 12:55:09 +0000)
Subject: Re: emacs mode for orc files etc
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>>>>> "Michael" == Michael Coble writes:
Michael> rasmus ekman wrote:
>>
>> Jonas B|low wrote:
>> >
>> > I read about a emacs mode for editing csound orc files. Where can I
>> > find source code for this mode?
>>
>> ftp://ftp.maths.bath.ac.uk/pub/dream/utilities/Emacs-macros/
>>
Michael> I downloaded these emacs lisp files, attempted to byte-compile them, and they
Michael> were referencing another file "tempo." Where is tempo.el? Another location perhaps?
Strangely enough it is at
ftp://ftp.maths.bath.ac.uk/pub/dream/utilities/Emacs-macros/tempo.el
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To: csound@maths.ex.ac.uk
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From: tolve
Subject: realtime csound
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couple things rattling inside my head regarding strategies that just might
help a few of those on the borderline achieve better realtime. have not
seen posted.
when working with recording software, the rule seems to be to store
soundfiles on a hard drive separate from that which houses system and
application. for csound instruments that call on soundfiles, and an
otherwise almost fast enough processor, maybe storing such soundfiles on an
extra hard drive could make a difference? of course this would be a cheap
fix only for those already in possession of an additional drive.
on mac, what about loading said soundfiles into ram disk? hmm or try even
csound application in ram disk. imagine there are similar alternatives for
other platforms.
tolve
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From: Jim Stevenson
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To: csound@maths.ex.ac.uk, tlv@tuna.net
Subject: Re: realtime csound
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Which does it better in quad, an SGI or Sun, or another work station?
Are these faster than the fastest PC with which 4 chanel sound card?
Thanks.
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Subject: Re: Floating point problems and sscanf
To: Csound mailing list
Date: Tue, 7 Apr 1998 12:13:21 -0400 (EDT)
From: Eli Brandt
In-Reply-To: <199804071042.LAA29597@music.gla.ac.uk> from "Stephen Brandon - SysAdmin" at Apr 7, 98 11:42:28 am
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Stephen Brandon - SysAdmin wrote:
> sr = 22050.000000, kr = 1000.000000, ksmps = 22.049999
> error: inconsistent sr, kr, ksmps
>
> Of course, I had tried to use ksmps as 22.05.
>
> At first I thought it was the sscanf in otran.c (constndx) that was
> playing up, but then I experimented with some code fragments only
> to find that even if I assigned "(float) newval = 22.05", "if
> (newval == 22.05)" would fail.
As you probably know, 22.05 can't be represented exactly. I'm not
certain what's going on in your code fragment (I blindly avoid all
equality tests on floating-point, myself), but I imagine newval is
being promoted to a double and compared against a longer approximation
of 22.05.
The simplest approach is to avoid assigning to all three of these
values. Presuming you need the third, you could just call it
something else that Csound won't try to check for consistency.
(Might "ksmps = sr/kr" work better? Dunno whether it's even legal.)
When Csound is given two of the three, it computes the third internally --
might want to make it available to the .orc. (If it already does this,
well, good.)
--
Eli Brandt | eli+@cs.cmu.edu | http://www.cs.cmu.edu/~eli/
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From: "Vercoe, Scotty"
To: CSOUND
Subject: ADI Extended Csound news
Date: Tue, 7 Apr 1998 12:14:52 -0400
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Hi everyone,
As many of you know, Analog Devices is working to bring Csound to wider
audience, providing greater MIPS-per-buck as host-based systems. As a
member of this team, I watch this list, and am happy to answer any
questions anyone may have.
I am first and foremost a musician, graduating in 1995 from Oberlin with
a B.M. in jazz composition. My father mentioned that ADI needed someone
to make MIDI files, play keyboard for demos, edit samples and keep track
of libraries, which was fine, despite my lack of experience with Csound.
I was very lucky to have a mentor like Richard Boulanger, and I got more
and more into making instruments.
In June 1997, our project manager, Lee Ray, decided to pursue other
interests. At the last minute, everyone pointed to me, and although I
had no interest in management/marketing, there was nobody else, and I
couldn't just let things fall apart. Trying to pick up the very
scattered pieces that were left for me, I tried to redefine what we
needed. I brought on Jean Piche, Alex Burton and later Michael Gogins,
adding to our existing consultant team of Russell Pinkston and John
ffitch.
We started shipping our SDK 1.0 beta release last October. We were very
happy to finally get this out the door and into people's hands. I am
not going to lie, there are some holes in the system, and the driver
needs a little work, but many are finding it to be a truly workable and
exciting development environment. Keep in mind that Analog Devices is a
chip company, so doing software alone is a stretch! Our Software and
Systems Technology Division led by Mike Haidar, is dedicated to
enhancing the value of ADI's chips with software for dedicated
applications. Writing drivers for multiple platforms does not make
sense because we are not trying to make money off the development kit.
The goal of any kit from ADI is to provide customers and developers with
a system they can use to prototype products that use our chips.
I realize that there has been a lack of information on the project, and
many of you are left wondering. For this, I am sorry. Although
Analog's commitment to this project is definitely assured, we are a very
small hardworking team, and I constantly wish things could move faster
and see things that must be put off due to time and resource
constraints.
So...how can we get to the point of bringing an orchestra file to a gig
and load it on your friend's Csound synth? If this seems like the
future you want to be part of, your input is invaluable. If you want
Mac drivers, pressure your favorite Mac companies like MOTU or Opcode to
start making cards; if you want to see a synth, pressure some synth
companies. Mike Haidar does not subscribe to this list, but, at some
personal risk, I encourage you all to let him know how much revenue this
project could generate. His email is:
Mike.Haidar@analog.com
I am doing the best I can in the situation. If you have any advice on
what I should be doing (or moral support) I welcome all input. I can
assure you that my father and all members of this team are working
incredibly hard to make this project succeed.
Thanks,
Scotty
P.S. I will be at AES/Amsterdam in May, so come by the Analog Devices
booth and say hi!
Scotty Vercoe
Extended Csound Applications Consultant
Analog Devices Software & Systems Technology Division
Tel: (781) 461-3569 FAX: (781) 461-4291
Support: Csound.support@analog.com
Website: http://www.analog.com/
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Date: Tue, 07 Apr 1998 18:48:14 +0200
From: khalid
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To: Nicola Bernardini
Cc: csound@maths.ex.ac.uk, linux-audio-dev@ginette.musique.umontreal.ca
Subject: Re: [linux-audio-dev] Re: ADI Csound SDK/Csounders swear !?!
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Nicola Bernardini wrote:
>
>
> Why? If the specs where public, it would probably be so quick to
> build the *right* development tools for that card; oh well, who cares,
> there are so many other cards...
>
> Nicola
I haven't noticed any sign that ADI is actually aiming at developing a
PCI card for the market. I guess they make these cards for developement
of apps that would then run in embedded systems. If you consider how
many people (buyers) actually care bout things like "LPC" or grain-
synthesis you might understand them. We software synthesis lovers are
still a minority that won't grow so quickly, I believe.
Meanwhile, you could have a look at the MAAS/Highend-sound-driver pro-
ject. This board has a Motorola 56301 DSP on it. It is said to reach
80.000 MFlops (the SHARC DSP is about 120.000). There are development
tools for the 56k family running under Linux (though I haven't used
them). B.t.w. - great work with the parser!!
kd
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Date: Tue, 7 Apr 1998 17:19:19 -0400 (EDT)
From: Piche Jean
Subject: Re: realtime csound
To: Jim Stevenson
Cc: csound@maths.ex.ac.uk, tlv@tuna.net
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>
> Which does it better in quad, an SGI or Sun, or another work station?
> Are these faster than the fastest PC with which 4 chanel sound card?
>
I think the sgi is the only workstation with quad output (and thats the
Indy only -- O2 has 8 output with the RAD card)... I'll venture that the
sgi is among the faster machines but thta is slowly being eroded by newer
Pentium(a) and G3s (but these will not in general do multi-track audio
output)...
_____________________________________________________________________________
Jean Piche
Musique - UdM
pichej@ERE.Umontreal.ca
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Subject: Re: realtime csound
To: Csound mailing list
Date: Tue, 7 Apr 1998 18:48:12 -0400 (EDT)
From: Eli Brandt
In-Reply-To: from "Piche Jean" at Apr 7, 98 05:19:19 pm
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Piche Jean wrote:
> Pentium(a) and G3s (but these will not in general do multi-track audio
> output)...
For the PC there are plenty of multi-out cards; somewhat fewer multi-ins.
I think they show up under Windows as multiple stereo devices, but
with synchronized sample clocks. Unfortunately there's spotty NT support,
and I don't know of any for Linux (but haven't looked recently).
http://www.xs4all.nl/~rexbo/pc_hwsw.htm
--
Eli Brandt | eli+@cs.cmu.edu | http://www.cs.cmu.edu/~eli/
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Date: Wed, 08 Apr 1998 01:04:23 +0200
From: rasmus ekman
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Eli Brandt wrote:
>
> Stephen Brandon - SysAdmin wrote:
> > sr = 22050.000000, kr = 1000.000000, ksmps = 22.049999
>
> As you probably know, 22.05 can't be represented exactly. I'm not
> certain what's going on in your code fragment
Um, let it be known that ksmps must be integer, ok?
eg sr = 22050, kr = 294, ksmps = 75, etc.
(If you're poking around in the code, you'll find that
ksmps is used as a loop counter: control variables are
updated at the lower k-rate (kr), then each instrument
computes ksmps number of samples using these values.)
> (Might "ksmps = sr/kr" work better? Dunno whether it's even legal.)
Should be, but as you say, unnecessary.
hth,
re
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jpff@maths.bath.ac.uk wrote:
>
> I originlly wrote the heartbeat function to use . characters, but i
> was prevailed upon to produce the rotating bar. Is there a consensus,
> or do you want this as a controlled option? Advice please.
I'd like number of buffers written, followed by time:
nbufs (secs)
- as a new option, say command line flag "-H1".
Keep the rotating bar as "-H" and "-H0".
A tall order for a marginal comfort...?
rasmus
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jpff@maths.bath.ac.uk wrote:
>
> I originlly wrote the heartbeat function to use . characters, but i
> was prevailed upon to produce the rotating bar. Is there a consensus,
> or do you want this as a controlled option? Advice please.
I'd like number of buffers written, followed by time:
nbufs (secs)
- as a new option, say command line flag "-H1".
Keep the rotating bar as "-H" and "-H0".
A tall order for a marginal comfort...?
rasmus
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Subject: Where's the source? ( for NT )
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Howdy. I'm new to the list, but have pored over every csound WWW link and
ftp site I can find and can't seem to find source code anywhere.
Am I overlooking something at the Bath site? I've noted a message from
John Fitch back in Jan or Feb that he was having trouble uploading the
source, so perhaps it isn't available yet? (for ver3.47)
Thanks for any help.
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From: Matt
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> jpff@maths.bath.ac.uk wrote:
> > I originlly wrote the heartbeat function to use . characters, but i
> > was prevailed upon to produce the rotating bar. Is there a consensus,
> > or do you want this as a controlled option? Advice please.
I do a fair bit of stuff with a single long note, single instrument,
which means I generally get one message then a long pause then a "x bytes
written to whatever" message. I'd really like something along the
lines of
30.5/360.0 seconds
etc. Thanks.
My 2 cents worth :)
Matt
----------------------------------------------------------------------------
Matt Southall - AARG http://www.psyc.nott.ac.uk/research/aarg
AI Group, Dept. of Psychology, University of Nottingham, Nottingham NG7 2RD.
email: mjs@psychology.nottingham.ac.uk, tel: +44 (0)115951 5151 ext. 8311
----------------------------------------------------------------------------
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Date: Tue, 07 Apr 1998 20:31:46 -0500
From: pete moss
Organization: pete moss GmbH
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To: csound
Subject: digeridoo
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i am interested in creating a digerdoo-like instrument. anyone have any
ideas? i have tried gran synth of samples but so far that sounds like
poop. i need something more realistic, but havent a clue as to how to
approach this one...
pete
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From: Dustin Barlow
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To: Jim Stevenson
Cc: "csound@noether.ex.ac.uk"
Subject: RE: realtime csound
Date: Tue, 7 Apr 1998 20:37:24 -0500
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I think Windows NT boxes are a great alternative to the typically high
priced SGIs and SUNs. One of the advantages of the NT operating system is
that it can support up to 4 Pentium processors (V 4.0), giving you lots of
"horsepower". NT5.0 may support between 6-8 Pentium processors...
Another nice thing is that if you can't afford to buy more than one
processor at a time, you can get a board that will support 4, and add them
as you can afford them.
There is not much, if any, commercial support for audio programs
specifically designed for NT because unfortunately, NT has only been widely
used in the corporate world. However, if you are familiar with low level
API calls, you can write some really powerful real-time synth apps that
utilize multiple audio cards....I've been working on a few myself...Most of
the commercial Win95 32-bit audio programs I've tried work just fine in NT
as well. No DirectX stuff though...:{
I do alot of dynamic web site development on NT boxes for a living and I
think that the future of BillySoft's Wintel Land is in the NT operating
system. The best bang for the buck is in the multiprocessor NT based
systems in my opinion....this is good news for those of us who can't go
work on hotdog university machines any more and can't afford to buy one for
ourselves.
Dustin Barlow
http://www.omnids.com
-----Original Message-----
From: Piche Jean [SMTP:pichej@ERE.UMontreal.CA]
Sent: Tuesday, April 07, 1998 4:19 PM
To: Jim Stevenson
Cc: csound@noether.ex.ac.uk; tlv@tuna.net
Subject: Re: realtime csound
>
> Which does it better in quad, an SGI or Sun, or another work station?
> Are these faster than the fastest PC with which 4 chanel sound card?
>
I think the sgi is the only workstation with quad output (and thats the
Indy only -- O2 has 8 output with the RAD card)... I'll venture that the
sgi is among the faster machines but thta is slowly being eroded by newer
Pentium(a) and G3s (but these will not in general do multi-track audio
output)...
________________________________________________________________________
_____
Jean
Piche
Musique -
UdM
pichej@ERE.Umontrea
l.ca
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Subject: Re: Floating point problems and sscanf
To: Csound mailing list
Date: Tue, 7 Apr 1998 21:49:52 -0400 (EDT)
From: Eli Brandt
In-Reply-To: <352AB0F7.3BFF@hexagon.se> from "rasmus ekman" at Apr 8, 98 01:04:23 am
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rasmus ekman wrote:
> Um, let it be known that ksmps must be integer, ok?
Oh, duh -- I got in "floating-point equality" mode and forgot what
ksmps _was_. :-)
--
Eli Brandt | eli+@cs.cmu.edu | http://www.cs.cmu.edu/~eli/
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From: Hans Mikelson
To: csound
Subject: Re: digeridoo
Date: Tue, 7 Apr 1998 21:48:41 -0500
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-----Original Message-----
From: pete moss
To: csound
Date: Tuesday, April 07, 1998 8:37 PM
Subject: digeridoo
Hi,
>i am interested in creating a digerdoo-like instrument. anyone have any
>ideas
I think you'll want to use a digital waveguide model but I've never tried
this. The sound is generated by keeping your lips pretty loose and
vibrating them into a tube ~4 cm in diameter and 1 meter long. You can use
PVC electrical conduit or plastic plumbing pipe or anything else you have
around. The characteristic sound, somewhat reminiscent of a resonant filter
sweep, is created by puffing out your cheeks and letting them go down again.
I sometimes practice on some pipe I found in the garage. Actually it may be
easier to build and learn to play one than it is to synthesize it. You can
find building instructions on the net.
Bye,
Hans Mikelson
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From: Paul Winkler
To: csound@maths.ex.ac.uk, kaeru@flash.net
Subject: Re: digeridoo
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>Date: Tue, 07 Apr 1998 20:31:46 -0500
>From: pete moss
>To: csound
>Subject: digeridoo
>
>i am interested in creating a digerdoo-like instrument. anyone have
any
>ideas?
Hmmm... various noise sources through a heavily resonant comb filter?
Control the drone pitch with the res. freq. of the comb filter, then mix
in parallel a pretty narrow bandbass filter which you sweep to do the
overtone thing...
might not sound anything like a didgeridoo, but I think I'll try
building it anyway. :)
PW
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Date: Tue, 7 Apr 1998 21:35:33 -0700 (PDT)
From: Wayne Freno
To: Csound List
Subject: Comb Reverb (newbie question)
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Hello list (and the people on it :-)
I'm a newbie so please forgive the following question,
which I'm quite sure probably has a simple solution. It
has to do with scaling amplitudes.
When I turn on comb reverb, the first two seconds of it
are always out of range, even if there are no other signals
present. After the reverb is on and then I turn the notes on,
everything is in range. Thank you for any help. (If other
newbies want to email me privately, perhaps we can help each
other out and not bother the big wigs around here ;-)
; BEGIN ORCHESTRA
sr = 44100
kr = 4410
ksmps = 10
nchnls = 2
ga1 init 0
instr 1
iscale = .5
ipan = p4
k2 linseg 1, 2, 10, 2, 20, 2, 30, 2, 40
k1 linseg 1000, 2, 800, 2, 600, 2, 400, 2, 200
asig1 foscil 15000*iscale, k1, 2, k2, 2, 1
asig2 foscil 15000*iscale, k1, 2, k2, 2, 1
asig3 = asig1 + asig2
outs asig3 * ipan, asig3 * (1 - ipan)
ga1 = ga1 + (asig3 * .2) ; reverb gain
endin
instr 99
a3 comb ga1, 2.5, .1
outs a3, a3
ga1 = 0
endin
; BEGIN SCORE
f1 0 4096 10 1
;instr strt dur pan
i1 0 2 0 ; play 2 notes w/o reverb
i1 3 2 1
i99 5 10 ; turn on reverb
i1 5 2 1 ; play 4 notes w/ reverb
i1 7 2 .7
i1 9 2 .3
i1 11 2 0
e
end soundbite
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From: David Boothe
To: 'Wayne Freno'
Cc: "'csound@maths.ex.ac.uk'"
Subject: RE: Comb Reverb (newbie question)
Date: Wed, 8 Apr 1998 01:28:32 -0500
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> -----Original Message-----
> From: Wayne Freno [SMTP:wfreno@vcn.bc.ca]
> Sent: April 07, 1998 23:36
> To: Csound List
> Subject: Comb Reverb (newbie question)
>
>
> When I turn on comb reverb, the first two seconds of it
> are always out of range, even if there are no other signals
> present. After the reverb is on and then I turn the notes on,
> everything is in range.
[David Boothe]
Sending ga1 back into itself:
ga1 = ga1 + (asig3 * .2)
creates a feedback loop. Use separate paths for comb's input and output.
Clear both after each use.
;REVISED ORC--------------------------------------
sr = 44100
kr = 4410
ksmps = 10
nchnls = 2
ga1 init 0
ga2 init 0
instr 1
iscale = .5
ipan = p4
k2 linseg 1, 2, 10, 2, 20, 2, 30, 2, 40
k1 linseg 1000, 2, 800, 2, 600, 2, 400, 2, 200
asig1 foscil 15000*iscale, k1, 2, k2, 2, 1
asig2 foscil 15000*iscale, k1, 2, k2, 2, 1
asig3 = asig1 + asig2
outs asig3 * ipan, asig3 * (1 - ipan)
ga1 = ga2 + (asig3 * .2) ; reverb input signal (i.e. aux
send on mixer)
endin
instr 99
ga2 comb ga1, 2.5, .1 ;generate reverb
outs ga2, ga2 ;output reverb
ga2 = 0 ;cleanliness is next to
godliness
ga1 = 0 ; (also get rid of pesky
feedback)
endin
> (If other newbies want to email me privately, perhaps we can help each
>
> other out and not bother the big wigs around here ;-)
[David Boothe]
That's one reason this list exists, although I don't qualify as
a big wig.
-dB
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