| Hi,
Here's a Saturday treat for you. Record a stereo 44.1 k sound file, name it
soundin.11 and save it in your sound input directory for use with the
following orchestra & score. I use readings from science fiction stories
myself. Basically this orchestra uses a resonant low-pass filter and
replaces the resonance oscilation with noise bands or frequency modulation
oscilator.
Have fun,
Hans Mikelson
; ORCHESTRA
; Coded by Hans Mikelson 5/23/98
; 1. Noise resonance
; 2. FM resonance
sr=44100
kr=4410
ksmps=10
nchnls=2
;---------------------------------------------------------------------------
; Sample with Noise Filter Resonance
;---------------------------------------------------------------------------
instr 1
idur = p3 ; Duration
iamp = p4 ; Amplitude
irate = p5 ; Read rate
isndin = p6 ; Sound input file
ipantab = p7 ; Pan table
iband = 0 ; Just leave it zero...
ifqcadj = .149659863*sr ; Adjustment for filter frequency
kfco expseg 800, idur/5, 5000, idur/5, 1000, idur/5, 2000, idur/5, 700,
idur/5, 3000 ; Freq Cut-Off
krez = 16 ; Resonance level
ain1, ain2 diskin isndin, irate ; Read sound file
axn = ain1+ain2 ; Mix the two together
; Resonant Lowpass Filter (4 Pole)
kc = ifqcadj/kfco ; Filter
constant
krez2 = krez/(1+exp(kfco/11000)) ; Adjust for
high frequencies
ka1 = kc/krez2-(1+krez2*iband) ; Adjust for
band pass character
kasq = kc*kc ; c^2
kb = 1+ka1+kasq ; Find b
ayn nlfilt axn/kb, (ka1+2*kasq)/kb, -kasq/kb, 0, 0, 1 ; Use the
non-linear filter linearly
ayn2 nlfilt ayn/kb, (ka1+2*kasq)/kb, -kasq/kb, 0, 0, 1 ; 4-Pole
; Resonant Lowpass Filter (4 Pole)
kcl = ifqcadj/kfco
krez2l = 2.0/(1+exp(kfco/11000)) ; Same as above
but low resonance this time.
ka1l = kcl/krez2l-(1+krez2l*iband)
kasql = kcl*kcl
kbl = 1+ka1l+kasql
aynl nlfilt axn/kbl, (ka1l+2*kasql)/kbl, -kasql/kbl, 0, 0, 1
ayn2l nlfilt aynl/kbl, (ka1l+2*kasql)/kbl, -kasql/kbl, 0, 0, 1
ares1 = ayn2-ayn2l ; Isolate the resonance
krms rms ares1, 100 ; Compute the average resonance
level
arand1 rand krms ; Noise source
ares2 butterbp arand1, kfco, kfco/4 ; Band filter it based on
filter cut-off
ares3 butterbp arand1, kfco*2, kfco/4 ; Get another band
aout = ayn2l+(ares2+ares3) ; Add back on to the low-pass
signal
kpan oscil 1, 1/idur, ipantab ; Panning
kpanl = sqrt(kpan)*iamp
kpanr = sqrt(1-kpan)*iamp
outs aout*kpanl, aout*kpanr ; Give it to me.
endin
;---------------------------------------------------------------------------
; Sample with FM Filter Resonance
;---------------------------------------------------------------------------
instr 2
idur = p3 ; Duration
iamp = p4 ; Amplitude
irate = p5 ; Read rate
isndin = p6 ; Sound input file number
ipantab = p7 ; Panning table
iband = 0 ; Controls band-pass character of the
filter but
ifqcadj = .149659863*sr ; values other than zero make the
filter unstable
kfco expseg 5000, idur/5, 1000, idur/5, 2000, idur/5, 700, idur/5, 3000,
idur/5, 800 ; Filter sweep
krez = 16 ; Resonance
ain1, ain2 diskin isndin, irate ; Read in the sound file (must be
stereo)
axn = ain1+ain2 ; Just mix together
; Resonant Lowpass Filter (4 Pole)
kc = ifqcadj/kfco ; Find the
mysterious constant "C"
krez2 = krez/(1+exp(kfco/11000)) ; Reduce res
at high frequencies
ka1 = kc/krez2-(1+krez2*iband) ; Compute A1
kasq = kc*kc ; C^2
kb = 1+ka1+kasq ; Find B
ayn nlfilt axn/kb, (ka1+2*kasq)/kb, -kasq/kb, 0, 0, 1 ; Use nlfilt
for filter sweep.
ayn2 nlfilt ayn/kb, (ka1+2*kasq)/kb, -kasq/kb, 0, 0, 1 ; Do it again
for four pole.
; Resonant Lowpass Filter (4 Pole) ; Same as
above but low-pass without resonance.
kcl = ifqcadj/kfco
krez2l = 2.0/(1+exp(kfco/11000))
ka1l = kcl/krez2l-(1+krez2l*iband)
kasql = kcl*kcl
kbl = 1+ka1l+kasql
aynl nlfilt axn/kbl, (ka1l+2*kasql)/kbl, -kasql/kbl, 0, 0, 1
ayn2l nlfilt aynl/kbl, (ka1l+2*kasql)/kbl, -kasql/kbl, 0, 0, 1
ares1 = ayn2-ayn2l ; Extract
resonance
krms rms ares1, 100 ; How big is the
resonance
; Amp Fqc Car Mod MAmp Wave
ares2 foscil krms, kfco, 1, .75, 3, 1 ; Generate an FM
signal based on resonance
aout = ayn2l+ares2/4 ; Add it on to
the low-pass version
kpan oscil 1, 1/idur, ipantab ; Handle equal
power panning
kpanl = sqrt(kpan)*iamp
kpanr = sqrt(1-kpan)*iamp
outs aout*kpanl, aout*kpanr ; Output
endin
; SCORE
; Coded by Hans Mikelson 5/23/98
f1 0 8192 10 1 ; Sine wave
f10 0 1024 7 0 1024 1 ; Pan RL
f11 0 1024 7 1 1024 0 ; Pan LR
; Sta Dur Amp Pitch SoundIn Pan
i1 0.0 8.873 1.5 0.98 11 11
i2 0.17 8.525 1.5 1.02 11 10
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Subject: amplitude envelopes /practicality
Date: Sat, 23 May 98 20:46:05 +0000
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From: Drew Skyfyre
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Hello everyone,
Do I have to specify an individual envelope for each individual
note/chord ? Or is there an easier way, i.e. someway of writing some
intelligence into the orch. to deal with envelope settings depending on
one variable to be specified in the score,sort of like a "velocity" value
?
I'm having a lot of trouble dealing with notes getting chopped off by
succeeding notes and having to go about tweaking envelope settings.Too
tedious for extended use.
Can anyone offer some suggestions and insight into this subject ?
Regards,
Drew
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Date: Sat, 23 May 1998 16:29:41 +0100
To: csound@maths.ex.ac.uk
From: Dave Perry
Subject: SoundIN paths
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What is the standard format for the filepath string in a soundin or diskin
statement? In newer version's of CSound for the PC (I refer in particular
to Gabriel Maldonado's new version of RTSound) it seems to use a unix
seperator (/) and in others the MSDOS (\). Error's are produced if you use
the wrong type.
Any help greatly appreciated,
Dave Perry
BTW, does anyone know any win95 API calls which identify the number of
channels in a wave file? It's been bugging me.
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From: Toby
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To: "Juergen A. Erhard"
Cc: CSound
Subject: Re: Durations and Sampling Rate
References: <199805221217.OAA05309@fortschritt.laden.ilk.de>
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What is happening here is that you are changing
the sampling rate in your Csound orc, while this
information is not getting to the DAC. It remains
playing at some fixed rate. So if you make more
samples per theoretical second of a piece, the DAC
takes longer to spit them out, and thus lengthens
the piece, and lowers all of the pitches.
If you are using a Linux system, I recommend 'vplay'
as the command line sound file player. If you choose
'-W' on the Csound command line to be sure that you
are generating a 'wave' file, then vplay will pick
up the sampling rate from the header of the sound
file, and pass that information on to the Dac, so that
the pitches and durations of the sound remain constant.
Then you will want to choose the highest supported
sampling rate where sound quality is the most important,
and disk space and compile time are the least important.
As the latter two conciderations increase in priority,
the sampling rate must be lowered, sacrificing some of
the quality of the sound, most notibly high-end
frequency response.
Toby
-There otta be a law-
> But this seems to depend (even in the latest version (3.781)) on the
> sampling rate. For example, when I do the little example score in the
> manual, but set a sampling rate of 44.1kHz, the whole `piece' is much
> longer than 2.5 seconds (seems to me it's about 4.41 times as long...)
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From: Toby
Organization: Other than 12-tone
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To: "Juergen A. Erhard"
Cc: CSound
Subject: Re: Durations and Sampling Rate
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What is happening here is that you are changing
the sampling rate in your Csound orc, while this
information is not getting to the DAC. It remains
playing at some fixed rate. So if you make more
samples per theoretical second of a piece, the DAC
takes longer to spit them out, and thus lengthens
the piece, and lowers all of the pitches.
If you are using a Linux system, I recommend 'vplay'
as the command line sound file player. If you choose
'-W' on the Csound command line to be sure that you
are generating a 'wave' file, then vplay will pick
up the sampling rate from the header of the sound
file, and pass that information on to the Dac, so that
the pitches and durations of the sound remain constant.
Then you will want to choose the highest supported
sampling rate where sound quality is the most important,
and disk space and compile time are the least important.
As the latter two conciderations increase in priority,
the sampling rate must be lowered, sacrificing some of
the quality of the sound, most notibly high-end
frequency response.
Toby
-There otta be a law-
> But this seems to depend (even in the latest version (3.781)) on the
> sampling rate. For example, when I do the little example score in the
> manual, but set a sampling rate of 44.1kHz, the whole `piece' is much
> longer than 2.5 seconds (seems to me it's about 4.41 times as long...)
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From: david first
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Hello Everybody -
I am totally stumped by the situation below. I am trying to create a simple
extended tone that moves through various glissandi over its length. I seem to
have hit some type of barrier, however. Example1 runs fine. Example2 - which
includes just one more segment - will not run. I don't get an error message,
the perf just quits. As an experiment (to see if the limit was about the
amount of characters), I created example3. This too runs fine. So it's about
something else. Is there an ultimate limit to the number of segments in
linseg? Some of you may recall a similar problem I was having a couple of
months ago. At that time I was getting error messages regarding argspace. This
was remedied (as a couple of people suggested) through the addition of a large
amount of semi-colons to my orc. So that's not the problem either. Sorry if
this is long, but it seemed the best way to clearly demonstrate the situation.
I'm running v3.47 PPC (Mills version) w/Cecilia (though I don't imagine that's
part of the issue).
Thanks for any and all suggestions,
d1st
--------------------------------------------------------------------------------------
sr = 44100
kr = 4410
ksmps = 10
nchnls = 2
instr 2 ;example1 (runs fine)
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg p5*6/5, p3*3/512, p5*6/5, p3*1/512,
p5*4/3,p3*3/512,p5*4/3,p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512,p5*6/5, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512,
p5*9/8,p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512,p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512, p5*6/5,p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512,p5*9/8, p3*1/512, p5*4/3, p3*3/512, p5*4/3, p3*1/512,p5*6/5,
p3*7/512, p5*6/5, p3*1/512, p5*9/8,p3*3/512, p5*9/8, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*7/512, p5*6/5, p3*1/512,p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*15/512,p5*6/5, p3*1/512, p5*6/5,
p3*3/512, p5*6/5,p3*1/512, p5*4/3, p3*3/512, p5*4/3, p3*1/512,
p5*6/5,p3*3/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512,p5*9/8, p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512,p5*4/3, p3*3/512, p5*4/3, p3*1/512,
p5*6/5,p3*3/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8,p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512,p5*4/3, p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*3/512,p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8,p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512, p5*4/3,p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*3/512,p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8, p3*1/512,p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5,p3*7/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8,p3*1/512, p5*4/3, p3*3/512, p5*4/3,p3*1/512, p5*6/5,
p3*7/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8, p3*1/512,p5*4/3,
p3*3/512,p5*4/3, p3*1/512, p5*6/5, p3*7/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8,p3*1/512, p5*4/3,p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*7/512, p5*6/5, p3*1/512,p5*9/8,p3*3/512, p5*9/8,p3*1/512,p5*4/3
aout1 oscil kamp1, kcps, 1
aout2 oscil kamp2, kcps, 1
outs aout1, aout2
endin
-------------------------------
instr 2 ;example2 (stops the perf)
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg p5*6/5, p3*3/512, p5*6/5, p3*1/512,
p5*4/3,p3*3/512,p5*4/3,p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512,p5*6/5, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512,
p5*9/8,p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512,p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512, p5*6/5,p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512,p5*9/8, p3*1/512, p5*4/3, p3*3/512, p5*4/3, p3*1/512,p5*6/5,
p3*7/512, p5*6/5, p3*1/512, p5*9/8,p3*3/512, p5*9/8, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*7/512, p5*6/5, p3*1/512,p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*15/512,p5*6/5, p3*1/512, p5*6/5,
p3*3/512, p5*6/5,p3*1/512, p5*4/3, p3*3/512, p5*4/3, p3*1/512,
p5*6/5,p3*3/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512,p5*9/8, p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512,p5*4/3, p3*3/512, p5*4/3, p3*1/512,
p5*6/5,p3*3/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8,p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512,p5*4/3, p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*3/512,p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8,p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512, p5*4/3,p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*3/512,p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8, p3*1/512,p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5,p3*7/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8,p3*1/512, p5*4/3, p3*3/512, p5*4/3,p3*1/512, p5*6/5,
p3*7/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8, p3*1/512,p5*4/3,
p3*3/512,p5*4/3, p3*1/512, p5*6/5, p3*7/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8,p3*1/512, p5*4/3,p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*7/512, p5*6/5, p3*1/512,p5*9/8,p3*3/512, p5*9/8,p3*1/512,p5*4/3,p3*3/512,p5*4/3
aout1 oscil kamp1, kcps, 1
aout2 oscil kamp2, kcps, 1
outs aout1, aout2
endin
----------------------------------
instr 2 ;example3 (also runs fine)
kamp1 linseg p4, p3, p4
kamp2 linseg p4*0, p3, p4*0
kcps linseg p5*6/5, p3*3/512, p5*6/5, p3*1/512,
p5*4/3,p3*3/512,p5*4/3,p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512,p5*6/5, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512,
p5*9/8,p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512,p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*3/512, p5*6/5,p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*3/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512,p5*9/8, p3*1/512, p5*4/3, p3*3/512, p5*4/3, p3*1/512,p5*6/5,
p3*7/512, p5*6/5, p3*1/512, p5*9/8,p3*3/512, p5*9/8, p3*1/512, p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5, p3*7/512, p5*6/5, p3*1/512,p5*9/8,
p3*3/512, p5*9/8, p3*1/512, p5*6/5, p3*15/512,p5*6/5, p3*1/512, p5*6/5,
p3*3/512, p5*6/5,p3*1/512, p5*4/3, p3*3/512, p5*4/3, p3*1/512,
p5*6/5,p3*3/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512,p5*9/8, p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512,p5*4/3, p3*3/512, p5*4/3, p3*1/512,
p5*6/5,p3*3/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8,p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512,p5*4/3, p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*3/512,p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8,p3*1/512, p5*6/5,
p3*3/512, p5*6/5, p3*1/512, p5*4/3,p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*3/512,p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8, p3*1/512,p5*4/3,
p3*3/512, p5*4/3, p3*1/512, p5*6/5,p3*7/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8,p3*1/512, p5*4/3, p3*3/512, p5*4/3,p3*1/512, p5*6/5,
p3*7/512, p5*6/5, p3*1/512, p5*9/8, p3*3/512, p5*9/8, p3*1/512,p5*4/3,
p3*3/512,p5*4/3, p3*1/512, p5*6/5, p3*7/512, p5*6/5, p3*1/512, p5*9/8,
p3*3/512, p5*9/8,p3*1/512, p5*4/3,p3*3/512, p5*4/3, p3*1/512, p5*6/5,
p3*7/512, p5*6/5, p3*1/512,p5*9/8,p3*3/512, p5*9/8,p3*1/512,p5*400000000/300000000
aout1 oscil kamp1, kcps, 1
aout2 oscil kamp2, kcps, 1
outs aout1, aout2
endin
-----------------------------------------------------------------
f1 0 8192 10 1 .5 .333 .25 .20 .1667
t 0 100
i2 0 512 10000 195.5556
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From: David Vincelli
To: csound@maths.ex.ac.uk
Date: Sat, 23 May 1998 16:03:50 +0000
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Subject: score11?
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The Schindler Csound Tutorial makes use of score11 notation.
Would someone be so kind as to tell me how I turn this notation into
the notation I'm used to (the one used through out the csound
manual). It there a built in function to the csound executable that
will translate this? Or do I have no choice but to use my mind to
understand this (gasp I wouldn't want to dot that now would I.)
(btw, I'm on Win95 and Linux)
--
David Vincelli
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From: Mike Berry
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As a matter of course, I would put all of these segments into 2 tables and
read it with tablei. The first table contains the durations (as absolute note
time, not time intervals), the second the points. So:
ktswitch init 0
kindex init 0
kincrem init 0
istartcps table 0, 2
kcps init istartcps
kcurtime times
if ktswitch < kcurtime kgoto proc
kindex = kindex + 1
koldtime = ktswitch
ktswitch table kindex, 1
knewcps table kindex, 2
kincrem = (knewcps - kcps) / ((ktswitch - koldtime) * ksmps)
proc:
kcps = kcps + kincrem
f1 is the table of times
f2 is the table of frequencies
While ktswitch is less than the next time value, csound will jump over the
initialization stuff and just add the increment to kcps. Once the ktswitch
time value is reached, the next time value is read into ktswitch. The next
freq value is read into knewcps. Then kincrem is calculated to increment kcps
by the correct amount until it reaches knewcps just at the next time value. I
know it looks like a lot, but it allows you to keep all of those envelope
points in a table, which is a much better place for them. Just use gen02 to
create the tables.
--
Mike Berry
mikeb@nmol.com
http://www.nmol.com/users/mikeb
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Date: Sat, 23 May 1998 18:49:37 -0400
To: "Ruston, Paul" ,
"'csound@maths.ex.ac.uk'"
From: QATJD
Subject: Re: new member
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Well, blow me down. I studied composition as an undergrad at Clark
University in Worcester MA with (among others) Brian Belet; who was himself
a student of James Tenney. I have heard some of Tenney's work and remember
liking it quite a bit.
A small but encouraging world of composition out there.
Enjoy Csound! It's not just a language; it's an ADVENTURE.
DQ
At 10:20 AM 5/22/98 -0400, Ruston, Paul wrote:
>Hello, I hope I'm sending this to the right address. I've been
>encouraged to submit some information about myself to the 'list'.
>
>I'm a masters student at York University in Toronto, Can. studying
>composition with James Tenney (if you don't know of him, find out, he's
>a remarkable person (but the net doesn't provide much)). My interests
>over the past couple of years have been in the limitations of equal
>temperament and western notation. Csound seems like it may be an ideal
>format for experimentations with tunings, etc., as the availability of
>microtonal instruments and musicians interested in performing microtonal
>works has been quite frustrating. I'm pretty much a computer idiot but
>am incouraged now that I've actually managed to get Csound running on
>windows and had some immediate success creating some simple orchestras
>and scores that sound as intended.
>
>While my primary concern was not synthesis, I am becoming increasingly
>interested and I think Csound will push me more to this end.
>
>Thanks for your time, I'll try not to ask alot of stupid questions.
>P. Ruston
>
>
>
thank you for your bandwidth
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