| To: ada@skyenet.net (A. Archias)
From: kim@headspace.com (Kim Cascone)
Subject: Re: Re; Sound Synth CDROM
> Does anyone know if the new Sound Synthesis CDROM by
>Martin Ross from Musicbook plus might be worth while? And
>perhaps help some of the Synthesis technique experimenting?
>It runs about $50.00
is there a URL for this?
KIM
>Yes there is; http://www.musicbookplus.com/books/fp147.htm
They say there are sound examples for the techniques with it.
My thought was to support experiments with Csound.
ANDY
Andy
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Date: Sun, 16 Nov 1997 16:18:34 +0100
From: Josep M* Comajuncosas
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To: David Densmore
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Subject: Re: orc/sco errors
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I=B4m afraid that attached files would=B4n work for all platforms and all
e-mail readers. The main problems when trying to compile a csound
orquestra are line wraps, which take not too much to find and fix.
Sometimes opcode names collision... so please try to use another Csound
version ...
as far as my TB clone is concerned... there should=B4n be more problems
than those related with line wraps...
David Densmore wrote:
> None of the orc/sco pairs I get from this list ever work. They
> are always full of errors.
>
> I just tried to run the TB-303 clone (II) from Josep Comajuncosas,
> but no luck.
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From: Josep M* Comajuncosas
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Subject: TB-303 clone (II)
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Mmmm... I=B4m posting again my TB303 emulator because I =B4ve not yet
prepared my page...
It has been improved a lot, specially the filter. It also includes a
sort of accent simulator, and the whole thing now sounds really acid ;-)
enjoy it!
PS. I hope you=B4ll receive the mail complete... otherwise ask me a
repost.
;ORC
sr =3D 44100
kr =3D 44100
;sr =3D 22050
;kr =3D 22050
ksmps =3D 1
instr 1 ; TB-303 emulator
; coded by Josep M=AA Comajuncosas
; init variables
ibpm =3D p14; 4/4 bars per minute
ieventdur =3D 15/ibpm
inotedur =3D ieventdur; or smaller...
icount init 0; sequence counter
idecaydur =3D inotedur
imindecay =3D (idecaydur<.2 ? .2 : idecaydur); set minimum decay to .2 or
inotedur
imaxfreq =3D 1000; max.filter cutoff freq. when ienvmod =3D 0
imaxsweep =3D 6500; sr/2... max.filter freq. at kenvmod & kaccent=3D 1
ipitch table 0,4; first note in the sequence
ipitch =3D cpspch(6 + ipitch/100)
ireson =3D 3; scale the resonance as you like (you can make the filter to
oscillate...)
imaxamp =3D 20000; maximum amplitude
idist =3D .015; 1 for ultra hardcore distortion ;-)
; raising idist also raises the resonance
kaccurve init 0
; twisting the knobs ...
kfco line p4, p3, p5
kres line p6, p3, p7
kenvmod line p8, p3, p9
kdecay line p10, p3, p11
kaccent line p12, p3, p13
;kfco randh .5, .9 , .65
;kfco =3D .5+kfco
;kfco port kfco, .5
;kres randh .5, .8, .32
;kres =3D .5+kres
;kres port kres, .5
;kenvmod randh .45, 1, .79
;kenvmod =3D .55 + kenvmod
;kenvmod port kenvmod, .5
;kdecay randh .5, .5, .74
;kdecay =3D .5+kdecay
;kdecay port kdecay, .5
;kaccent randh .5, .75, .31
;kaccent =3D .5+kaccent
;kaccent port kaccent, .5
start:
afdbk init 0
;pitch from the sequence
ippitch =3D ipitch
ipitch table ftlen(4)*frac(icount/ftlen(4)),4
ipitch =3D cpspch(6 + ipitch/100)
iremaining =3D icount*ieventdur/p3
print ipitch, iremaining
; slide detector; cannot slide from last note to the first one
;islide table ftlen(6)*frac(icount/ftlen(6)),6
;if islide =3D=3D 1:
;islidedur
; accent detector
iacc table ftlen(5)*frac(icount/ftlen(5)), 5
if iacc =3D=3D 0 goto noaccent
ienvdecay =3D 0; accented notes are the shortest ones
iremacc =3D i(kaccurve)
kaccurve oscil1i 0, 1, .4, 3
kaccurve =3D kaccurve+iremacc;successive accents cause hysterical raising
cutoff
goto next
noaccent:
kaccurve =3D 0; no accent & "discharges" accent curve
ienvdecay =3D i(kdecay)
next:
; two envelopes & pitch with portamento
kmeg expseg 1, imindecay+((inotedur-imindecay)*ienvdecay),
.1+.9*ienvdecay
kveg linen 1, .03, inotedur, .016; attack should be 4 ms. but ...
kpitch linseg ippitch, .06, ipitch
; reinit code
timout 0, ieventdur , contin
icount =3D icount + 1
reinit start
contin:
; amplitude envelope
kamp =3D .5*kveg*((1-i(kenvmod)) + kmeg*i(kenvmod)*(.5+.5*iacc*kaccent))
; filter envelope
ksweep =3D kveg * (imaxfreq +
(.7*kmeg+.3*kaccurve*kaccent)*kenvmod*(imaxsweep-imaxfreq))
kfco =3D 20 + kfco * ksweep; cutoff always greater than 20 Hz ...
;kfco =3D (kfco > sr/2 ? sr/2 : kfco); could be necessary
;generate bandlimited sawtooth wave
abuzz buzz kamp, kpitch, sr/(2*kpitch), 1 ,0;bandlimited pulse
asaw integ abuzz,0
asawdc atone asaw,1
;resonant 8-pole LPF
ainpt =3D asawdc - afdbk
alpf tone ainpt,kfco
alpf tone alpf,kfco
alpf tone alpf,kfco
alpf1 tone alpf,kfco
alpf tone alpf1,kfco
alpf tone alpf,kfco
alpf tone alpf,kfco
alpf2 tone alpf,kfco
abal1 balance alpf1,asawdc
abal2 balance alpf2,asawdc
afdbk =3D abal1-abal2
afdbk balance afdbk,asawdc*idist
afdbk tablei afdbk*(.05+kres*ireson), 7, 1, .5
; + resonance -> + distortion
aremovedc atone afdbk,1
out imaxamp*aremovedc
endin
;SCO
f1 0 16384 10 1
f2 0 1025 19 .5 .5 270 .5
f4 0 16 -2 0 12 0 0 3 0 0 0 0 15 12 0 6 15 1 4; sequencer (pitches
are 6.00 + p/100)
f5 0 16 -2 0 1 0 0 0 0 0 0 0 1 0 1 1 1 0 0; accent sequence
f3 0 8193 8 0 512 1 1024 1 512 .5 2048 .2 4096 0; accent curve
f7 0 1024 8 -.8 42 -.78 200 -.74 200 -.7 140 .7 200 .74 200 .78 42
.8
; f7 borrowed from H.Mikelson=B4s TB-303 emulator. Tnx!
;------------ KNOB POSITION : INITIAL AND FINAL VALUES FROM 0 TO
1--------------------
; cutoff freq resonance envelope mod. decay accent bpm
; 0 - 1 0 ~ 1 ~ .1 - 1 0 - 1 0 - 1 40-300
; start end st end st end st end st end
i1 0 6 .01 .3 .2 .2 .1 .4 .05 .8 0 0
80
i1 7 6 .95 1 .1 1 .8 1 .1 .01 0
1 70
i1 14 6 0 1 .5 1 .1 1 1 .05 1 1 60
i1 21 6 .5 1 .95 1 1 .9 .01 0 1
1 80
i1 28 6 0 1 .5 1 .1 1 .05 0 .5 1 120
;------------------------------END OF
SCORE-------------------------------------------
e
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Date: Sat, 08 Nov 1997 13:54:16
To: csound@maths.ex.ac.uk
From: Hans Mikelson
Subject: cps2pch Question & Chord Strumming Orc/Sco
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Hello,
In search of the lost chord? Try this chord strumming orchestra and score.
This may be a useful utility for trying out chord progressions & song
writing. While using it a question developed. How do you use cps2pch with
unequal temperament? I tried to use a 12 step Werkmeister III tuning and
could not get it to work. When I try I get the following:
7.00 is root
7.01 goes up one semitone
7.02 is the same pitch as 7.01
7.03 goes up another semitone
7.05 is the same pitch as 7.04
etc.
Anyway here is the strumming orchestra score although I would like to
experiment with other temperaments if I can figure out how to get it to work.
Have fun and good luck,
Hans Mikelson
; Orchestra
; This instrument is designed for strumming chords.
; Coded by Hans Mikelson 11/15/97
sr=44100
kr=22050
ksmps=2
nchnls=2
; Strummer Instrument
instr 1
idur = p3 ; Duration
iamp = p4 ; Amplitude
iroot = p5 ; Root of the chord
ichord = p6 ; The type of chord
iwave = p7 ; Wave form
ireps = p8 ; Number of strums
ienv = p9 ; Envelope shape
iphase = 1-p10 ; Time between individual picks
itemper = p11 ; Temperament
icarr = 1 ; Carrier
imodu = 3 ; Modulator
; First note of the chord
kamp1 oscil 1, ireps/idur, ienv ; Used for envelope
ichd1 table 0, ichord ; Read the chord table
ifqc1 cps2pch iroot+ichd1/100, 12 ; Compute the frequency
; Amp Fqc Car Mod Index Wave
asig1 foscil kamp1, ifqc1, icarr, imodu, kamp1*4, iwave ; Use a simple
FM tone
; Second note of the chord
kamp2 oscil 1, ireps/idur, ienv, iphase
ichd2 table 1, ichord
ifqc2 cps2pch iroot+ichd2/100, 12
asig2 foscil kamp2, ifqc2, icarr, imodu, kamp2*4, iwave
; Third note of the chord
kamp3 oscil 1, ireps/idur, ienv, iphase*2
ichd3 table 2, ichord
ifqc3 cps2pch iroot+ichd3/100, 12
asig3 foscil kamp3, ifqc3, icarr, imodu, kamp3*4, iwave
; Fourth note of the chord
kamp4 oscil 1, ireps/idur, ienv, iphase*3
ichd4 table 3, ichord
ifqc4 cps2pch iroot+ichd4/100, 12
asig4 foscil kamp4, ifqc4, icarr, imodu, kamp4*4, iwave
; Sum and output the happy result
aout = asig1+asig2+asig3+asig4
outs aout*iamp, aout*iamp
endin
; Equal Temperament Test
instr 2
idur = p3
iamp = p4
ipch = p5
iwave = p6
ienv = p7
itemper = p8
icarr = 1
imodu = 3
kamp1 oscil 1, 1/idur, ienv
ifqc1 cps2pch ipch, -4
; Amp Fqc Car Mod Index Wave
asig1 foscil kamp1, ifqc1, icarr, imodu, kamp1*4, iwave
aout = asig1
outs aout*iamp, aout*iamp
endin
f1 0 8192 10 1
f2 0 1024 7 1 1024 -1
f3 0 1024 7 0 16 1 368 .6 512 .4 128 0
; Temperaments I can't figure out how to get the temperaments to work.
f4 0 16 -2 1.000 1.067 1.121 1.196 1.273 1.344 1.422 1.496 1.600 1.703
1.798 1.903 2.000 ; Werkmeister III
f5 0 16 -2 1.000 1.059 1.122 1.189 1.260 1.335 1.414 1.498 1.587 1.682
1.782 1.888 2.000 ; Equal
f6 0 16 -2 1.000 2.000 3.000 4.000 1.260 1.335 1.414 1.498 1.587 1.682
1.782 1.888 2.000 ; Equal
; Chords
f10 0 4 -2 0 4 7 12 ; Major
f11 0 4 -2 0 3 7 12 ; Minor
f12 0 4 -2 0 5 7 12 ; Sus4
f13 0 4 -2 0 2 7 12 ; Sus2
f14 0 4 -2 0 4 7 10 ; 7th
f15 0 4 -2 0 3 7 10 ; Minor7th
f16 0 4 -2 0 3 7 11 ; Min/maj7th
f17 0 4 -2 0 4 7 11 ; Maj7th
f18 0 4 -2 0 4 7 9 ; Major6th
f19 0 4 -2 0 3 7 9 ; Minor6th
f20 0 4 -2 0 4 10 14 ; 9th
f21 0 4 -2 0 4 10 17 ; 11th
f22 0 4 -2 0 4 10 21 ; 13th
f23 0 4 -2 0 3 6 12 ;
;6/9: 0 4 7 9 14
;Min9th: 0 3 7 10 14
;Min11th: 0 3 7 10 17
;Min13th: 0 3 7 10 21
;Maj9th: 0 4 7 11 14
;Maj11th: 0 4 7 11 17
;Maj13th: 0 4 7 11 21
;add2: 0 2 4 7
; Sta Dur Amp Root Chord Wave Repeat Env Phase Temperament
i1 0 1.6 8000 7.02 10 1 2 3 .20 4
i1 + 1.6 . 7.04 11 . 2 . . .
i1 . 0.8 . 6.09 10 . 1 . . .
i1 . 1.6 . 7.07 10 . 2 . .10 .
i1 . 1.6 . 7.02 10 . 1 . .20 .
;
i1 . 0.8 . 7.02 10 . 1 . .05 .
i1 . 0.8 . 7.02 11 . 1 . .05 .
i1 . 0.8 . 7.02 12 . 1 . .00 .
i1 . 0.8 . 7.02 13 . 1 . .00 .
i1 . 0.8 . 7.02 23 . 1 . .05 .
i1 . 1.6 . 7.02 10 . 1 . .10 .
; Sta Dur Amp Pitch Wave Env Temperament
;i2 0 .2 8000 7.00 1 3 4
;i2 + . . 7.01 . . .
;i2 . . . 7.02 . . .
;i2 . . . 7.03 . . .
;i2 . . . 7.04 . . .
;i2 . . . 7.05 . . .
;i2 . . . 7.06 . . .
;i2 . . . 7.07 . . .
;i2 . . . 7.08 . . .
;i2 . . . 7.09 . . .
;i2 . . . 7.10 . . .
;i2 . . . 7.11 . . .
;i2 . . . 7.12 . . .
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Date: Sun, 16 Nov 1997 17:01:08 -0800 (PST)
From: Mike Berry
To: rasmus ekman
Cc: csound@maths.ex.ac.uk
Subject: Re: 32-bit float AIFF
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For the AIFF/C 32 float format you should contact Dave Madole
(madole@mills.edu) who wrote it and can pass it on.
Mike Berry
mikeb@mills.edu
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From: Richard Dobson
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Subject: Re: More than two channels
To: rasmus ekman
Date: Fri, 14 Nov 1997 20:28:53 +0000 (GMT)
Cc: csound@maths.ex.ac.uk
In-Reply-To: <346B8293.565A@hexagon.se> from "rasmus ekman" at Nov 13, 97 11:43:31 pm
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rasmus Ekman wrote re 4-ch playback program:
> Good idea. Should be no big problem; There are "number of channels"
> and "data block size" etc fields in the format header. I'm shure
> you could do it with a small to reasonable effort, but if you
> haven't studied the specification I could mail some references
> with explanations that I picked up over the last year or so
> (the spec has some curious quirks, and has been changed a while ago).
>
> Regards,
> re
>
I know about the standard WAV header stuff, as I already have a working
'standard' playback program, and I am hoping that extending to 4 channels
may be as simple as either:
1: opening a second WaveOut device and writing to two in parallel,
or
2: using a single device which itself allows 4 channels - if sucha device
exists!
I am also interested in supoporting the new WAVE_IEEE_FLOAT 32-bit format
which is described in Sonic Foundry's documentation for Active Movie plugins.
They supoport this format internally, in Sound Forge, but not to read or write
files, yet. I have modified our CDP sound filing system routines to
recognise this format, but I have no means of testing this until I find
a third-party program which also supports it at the file level. Any information or
help on this will be gratefully received!
Although 32bit float format soundfiles can be expected to be large, I can
imagine that the format may be useful to many Csound users, if only to import
floating-point normalized wavetables or whatever. Of course, this is already
possible in principle in Csound, but it would be good to provide explicit
support for the new WAV format.
Richard Dobson
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Date: Sun, 16 Nov 1997 00:11:30 +0100
From: rasmus ekman
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To: Mike Berry
Cc: csound@maths.ex.ac.uk
Subject: 32-bit float AIFF
References:
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Mike Berry wrote:
>
> In the PPC version of CSound we support a 32 bit float
> version of AIFF/C. It is extremely useful.
Could you please mail/post info on the header changes (I have the
basic AIFF spec). I'd like to play around with this format at home.
> Perhaps that format should be extended to all the platforms
Vigorously seconded. Would be nice not to spend half my CSound
time regenerating files with out-of-range samples.
regards,
re
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From: Mike Berry
To: Richard Dobson
Cc: rasmus ekman , csound@maths.ex.ac.uk
Subject: Re: More than two channels
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On Fri, 14 Nov 1997, Richard Dobson wrote:
>
> I am also interested in supoporting the new WAVE_IEEE_FLOAT 32-bit format
> which is described in Sonic Foundry's documentation for Active Movie plugins.
> They supoport this format internally, in Sound Forge, but not to read or write
> files, yet. I have modified our CDP sound filing system routines to
> recognise this format, but I have no means of testing this until I find
> a third-party program which also supports it at the file level. Any information or
> help on this will be gratefully received!
>
> Although 32bit float format soundfiles can be expected to be large, I can
> imagine that the format may be useful to many Csound users, if only to import
> floating-point normalized wavetables or whatever. Of course, this is already
> possible in principle in Csound, but it would be good to provide explicit
> support for the new WAV format.
>
> Richard Dobson
>
In the PPC version of CSound we support a 32 bit float version of
AIFF/C. It is extremely useful. Perhaps that format should be extended
to all the platforms, since AIFF is really the main cross-platform sound
file format. The peak value is stored in the header, so we even have
support to playback float files directly.
Mike Berry
mikeb@nmol.com
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From: Josep M* Comajuncosas
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Subject: POSCIL (release 3.47)
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I=B4m quite disapointed with the implementation of a variable pulse width
unit generator in the new 3.47 release.BV?
Apart from some amplitude related problems, I=B4m afraid the waveform is
generated by pure arithmetic methods, and the result is an extremely
aliased wave, which I find absolutely unnacceptable except, maybe, for
control signals. You can easily comfirm this by looking at a
spectrogram...
It is not difficult to generate bandlimited PWM... you can just use the
already implemented pulse generator (buzz) in that way:
sr =3D 44100
kr =3D 44100
ksmps =3D 1
instr 1; bandlimited PWM coded by Josep M=AA Comajuncosas
; note: you could further integrate the output
; to get Triangle waves with pulse modulation
kamp init 10000
kfreq line 100, p3, 200
krel line 1, p3 ,0; 0 to 1 (modulation index)
apulse1 buzz 1,kfreq, sr/(2*kfreq), 1;no aliasing here! ;-)
apulse delayr 1/50;don=B4t try to use below 50 Hz <- a better idea?
apulse2 deltapi krel/kfreq;interpolating to get the proper phase
delayw apulse1
avpw =3D apulse1 - apulse2; two inverted pulses at variable distance
apwmdc integ avpw; integrating gives a square wave with WM
apwm atone apwmdc,1; remove DC offset caused by integ
apwm balance apwm,apulse1; and restore original power
out kamp*apwm; look at its sonogram! it=B4s really nice!!
endin
f1 0 16384 10 1
i1 0 10
e
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To: csound@maths.ex.ac.uk
From: Javier Ruiz
Subject: 32 bit float AIFF
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And I really thank you for the time it saves.
P.S. I have just bought a 8600/200, should I buy more cache memory for real
time
performance? (I have already enough DRAM)
> In the PPC version of CSound we support a 32 bit float version of
>AIFF/C. It is extremely useful. Perhaps that format should be extended
>to all the platforms, since AIFF is really the main cross-platform sound
>file format. The peak value is stored in the header, so we even have
>support to playback float files directly.
>
>Mike Berry
>mikeb@nmol.com
Javier Ruiz
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Date: Sun, 16 Nov 1997 16:58:20 -0500 (EST)
From: David Densmore
To: csound@maths.ex.ac.uk
Subject: Re: orc/sco errors
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Toby Shepard wrote:
>David Densmore wrote:
>>
>>None of the orc/sco pairs I get from this list ever work. They
>>are always full of errors.
>>
>>I corrected all of the wraps I could see, but I still couldn't
>>get the TB-303 pair to execute properly.
>
>I had a similar feeling about posted orc's until I found out that
>the commonly used 'expseg' was broken on stock distributions
>compiled for Linux. I was frustrated about the 'bogus' posts,
>only to find out that it was a problem with my binary.
I run both Linux and Win 95 and I couldn't the TB-303 to execute
on either version.
Josep Comajuncosas mailed me the orc/sco pair as an attached zipfile
and they ran fine.
dd
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Date: Sun, 16 Nov 1997 19:32:50 +0100
From: Josep M Comajuncosas
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Im quite disapointed with the implementation of a variable pulse width
unit generator in the new 3.47 release.BV?
Apart from some amplitude related problems, Im afraid the waveform is
generated by pure arithmetic methods, and the result is an extremely
aliased wave, which I find absolutely unnacceptable except, maybe, for
control signals. You can easily comfirm this by looking at a
spectrogram...
It is not difficult to generate bandlimited PWM... you can just use the
already implemented pulse generator (buzz) in that way:
sr = 44100
kr = 44100
ksmps = 1
instr 1; bandlimited PWM coded by Josep M Comajuncosas
; note: you could further integrate the output
; to get Triangle waves with pulse modulation
kamp init 10000
kfreq line 100, p3, 200
krel line 1, p3 ,0; 0 to 1 (modulation index)
apulse1 buzz 1,kfreq, sr/(2*kfreq), 1;no aliasing here! ;-)
apulse delayr 1/50;dont try to use below 50 Hz <- a better idea?
apulse2 deltapi krel/kfreq;interpolating to get the proper phase
delayw apulse1
avpw = apulse1 - apulse2; two inverted pulses at variable distance
apwmdc integ avpw; integrating gives a square wave with WM
apwm atone apwmdc,1; remove DC offset caused by integ
apwm balance apwm,apulse1; and restore original power
out kamp*apwm; look at its sonogram! its really nice!!
endin
f1 0 16384 10 1
i1 0 10
e
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To: Jo Scherpenisse
From: Russell Pinkston
Subject: Re: turnoff statement (repost)
Cc: csound@maths.ex.ac.uk
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>Some weeks ago I have sent this question, but at that time I wasn't really
>on the csound list.
>Therefore I repeat the question:
>
>Is it possible to start a new instance of an instrument, say i1,
>exactly after a previous line in the scorefile (same instrument i1) stops
>because of a "turnoff" statement execution?
>The first i1 statement can have a negative p3, or a "hold" statement in the
>orc-file but will
>stop after a built-in condition is met. The time is not exactly predictable,
>but I want to start the next i1 line exactly at that moment.
>So I can not give that line a good p2 parameter.
>I hope someone has an idea about this. Thank you for any help.
>
>
>Jo Scherpenisse (hoofd elektronische dienst), e-mail: scherp@koncon.nl
>Koninklijk Conservatorium, Juliana van Stolberglaan 1
>2595 CA Den Haag tel: 070-3814251 fax: 070-3853941
>
>
[I'm reposting this, because it didn't seem to go out to the csound list on
my first attempt. RP]
I think you might be better off avoiding this whole problem. A different
approach would be to use reinitialization within the instrument. In other
words, if you're now issuing a turnoff when some conditions are met within
your instrument, issue a reinit command instead. That way, you don't have to
start a new note from the score each time you turn off the previous note;
you just use one i-statement for all the notes. Since (presumably) you're
now changing some parameters (like pitch, amp, etc.) for each new note via
the p-fields in the score, you would probably need to place those parameters
in a function table, instead, and reference them by table units, e.g.,
instr 1 ;simple reinit instr
iwavefn = 1 ;oscil function number
iparms = 2 ;parameter table number
index = 0 ;index into the parameter table
reinitfromhere: ;section to be reinitialized
ipch table index,iparms ;get this pitch from table
iamp table index+1,iparms ;get this amp from table
idur table index+2,iparms ;get this dur from table
index = index+3 ;update index for next note
kgate linen iamp,.01,idur,.1
asig oscil kgate,cpspch(ipch),iwavefn
out asig
timout 0,idur,skipinit ;skip over reinit for idur secs
reinit reinitfromhere ;reinit after idur
rireturn ;end of sect to be reinitialized
;N.B., rireturn is not really needed,
;if at end of instr
skipinit:
endin
;sample score
f01 0 2048 10 1
;function with parm data: pch amp dur
f02 0 16 -2 8.00 20000 .5 ;parms in sets of 3
8.02 10000 .5
8.04 5000 .5
8.05 10000 .5
8.07 20000 2
i01 0 4 ;other parms for entire phrase here
e
...or something like that. I'm using timout to determine the note lengths,
but you could use other methods to test for when to reinit.
Good luck,
Russell Pinkston
----------------------------------
Russell F. Pinkston, D.M.A.
Associate Professor of Composition
Director, Electronic Music Studios
School of Music
The University of Texas at Austin
Austin, TX 78712
[512-471-0865]
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Date: Sun, 16 Nov 1997 10:11:01 -0500 (EST)
From: David Densmore
To: csound@maths.ex.ac.uk
Subject: orc/sco errors
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None of the orc/sco pairs I get from this list ever work. They
are always full of errors.
I just tried to run the TB-303 clone (II) from Josep Comajuncosas,
but no luck.
Being new to Csound, it is difficult for me to find all the errors,
but one problem is obvious, line wraps.
In many instances, the last part of a line of code gets wrapped
to the next line, causing a syntax error. I don't know if this
is my fault or if it is something generic about email, but I
tried to get the message from the web archive and the line wrap
problem was present there too.
I corrected all of the wraps I could see, but I still couldn't
get the TB-303 pair to execute properly.
Does anyone else have this problem?
Perhaps orc/sco pairs should be zipped and sent to this list as
MIME attachments to prevent them from being corrupted via email
transit.
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Date: Mon, 17 Nov 1997 01:59:13 -0800 (PST)
From: "Matt J. Ingalls"
To: doug cross
Cc: arlen barr , Hans Mikelson ,
csound@maths.ex.ac.uk
Subject: Re: diskin question
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Doug,
i tested out your orc/sco and everything seemed to work fine for me except
for a one-sample-click at the beginning of every note. This seems to be
generated from the balance - i presume its because the reverb "will begin
to appear only after 1/20 second or so of delay" -- who knows what balance
is doing during that time (besides making a click)
easy fix to gate the output...
-matt
P.S. "diskin" has not been tested too much, there are a few existing bugs
dealing with backward looping and veryextreme pitch data --
but if anyone finds any problems if you could please report them, that
would be great!!!!
(thanks)
On Fri, 31 Oct 1997, doug cross wrote:
> hello everyone:
> just last night I discovered "diskin".
>
> I found a distortion when i tried to add reverb.
> I am puzzled. Any ideas?
> doug
> ;----------------------------temp.orc---------------------
>
> sr = 44100
> kr = 4410
> ksmps = 10
> nchnls = 2
>
> instr 1
>
> a1,a2 diskin "example.aiff", p4, 0, 1, 4 ;5.555374 sec my example files
> length
>
> ao1 reverb a1, 0.3, 1
> ao2 reverb a2, 0.3, 1
>
> abal1 balance ao1, a1
> abal2 balance ao2, a2
>
> outs abal1, abal2
>
> endin
> ;----------------------------temp.sco----------------------
>
> i1 0 5.555374 1 ; original transpose
> s
> i1 0 11.110748 .50 ; transpose b
> s
> i1 0 13.888435 .40 ; transpose c
> s
> i1 0 18.517913 .30 ; transpose d
> s
> i1 0 27.776870 .20 ; transpose e
> e
> ;----------------------------------------------------------------
>
>
>
>
|