| Josep M Comajuncosas wrote:
>
> rasmus ekman escribió:
>
> > ; kpitch, kamp pitch anysig, .2, 7.0, 8.0, 1, 12, 10, 7.5, 6, 10, 4, -1
>
> Hmm, do you have a working example of pitch? I strongly need it
> but I cannot make this opcode work.
No, I haven't tested it really. I just fed it random values to make
it compile (and demo the bugs).
Your problem is probably that there is a bug in the present code,
making it crash. I sent the fix to jpff; if anyone else wants it now,
here goes:
Line 257 in pitch.c should be moved up to line 254: The lines
if ((--p->scountdown)) goto nxt; /* if not yet time for new spec */
p->scountdown = p->timcount; /* else reset counter & proceed: */
downp = &p->downsig;
>>> specp = &p->wsig;
must be changed to:
>>> specp = &p->wsig;
if ((--p->scountdown)) goto nxt; /* if not yet time for new spec */
p->scountdown = p->timcount; /* else reset counter & proceed: */
downp = &p->downsig;
- else specp will have a garbage value, causing a crash when
it is used below the "nxt" label.
Cheers,
re
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From: patrick pagano
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Hi C sounders
I have been employing the section technique to remove unwanted residual
Hi frequencies in granny pieces and would like to attempt to use a
filter to further "clean" up the sections can someone suggest a nice
filter for this grain instr??
> instr 1
> ioutch = p12 ; Output channel
> ipanl = sqrt(p13) ; Pan left
> ipanr = sqrt(1-p13) ; Pan right
> k1 linen 1000,.1,p3,.1
> aamp linseg 0, .002, 1, p3-.004, 1, .002, 0 ; Declick envelope
> asig grain k1, p4, p5, p6, p7, p8, p9, p10,p11
> aout = asig*aamp
> outs aout*ipanl, aout*ipanr ; Output stereo
> zawm aout, ioutch ; Mix to the Zak channel
> endin
>
I would like to remove any extra hi freqs and any and all suggestions
are appreciated
Thanks
Pat
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From: Peter Neubacker
Date: Sun, 29 Aug 99 18:10:34 +0200
To: csound@maths.ex.ac.uk
Subject: Re: Phasor notes
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Sean Costello wrote:
> "phaser" would be too close to phasor to be useful, I think.
I think that's right, as the're seems to be some confusion
with it anyway...
Peter
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dze kl!ent : http://www.m9ndfukc.org/kode/k!berzveta||.sit.bin
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\ \ kontrl + _move kurzr = kreate nO!sz
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or 2.1 chekd - [1 l!ne]
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From: Hans Mikelson
To: Csound
Subject: Re: VCO Opcode
Date: Sun, 29 Aug 1999 15:26:34 -0500
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Hi,
Anders wrote:
>Why do I have to set KR=SR when using it?
>
>It seems that there is a bug somewhere that will make the opcode act very
>strange when using lower KR like 4-500Hz.
I did not intend this. If you send the offending orc/sco I will look at it
and see if I can tell what is happening. I was not able to create a problem
at krD1 with a simple example.
Bye,
Hans Mikelson
; ORC
srD100
krD1
ksmps0
nchnls=2
instr 1
iamp = p4
ifqc = cpspch(p5)
a1 vco iamp, ifqc, 2, .5, 1, 2/ifqc
outs a1, a1
endin
; SCO
f1 0 65536 10 1
i1 0 1 20000 6.00
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From: Sean Costello
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CC: rasmus ekman ,
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Subject: Re: New bug generator opcode
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Josep M Comajuncosas wrote:
>
> I had a similar idea to emulate subtle variations of analog equipment. Maybe it
> would be nice to define a new set of variables, say ?i,?k,?a variables, which
> could have an added (and user definable) amount of thermal noise in them. The
> most important think should be the kind of randomness to be applied, and where.
It seems like at least part of this thermal noise could be emulated with randi.
The drift of analog VCOs, for example, could be controlled by this. However,
maybe you want to have a global temperature variable, which affects all ugens
that are controlled by the ?i, ?k, or ?a variables. Changing the temperature
would cause all of the associated ugens to change their behavior, but in an
unpredictable way (i.e. all the oscillators would drift upwards in pitch, but at
different rates and different amounts).
I'm not sure how you would emulate the "jitter" of analog VCOs, as I don't think
that this has been studied in detail. Some VCOs (Minimoog, VCS3) exhibit a
certain amount of pitch "jitter" that undoubtedly has an effect on the
perception of the sound. Perhaps noise through a bandpass filter of relatively
narrow bandwidth could simulate this effect.
I'd say that the tricky part of emulating analog equipment is emulating the
nonlinearities. An analysis of the circuit that is being emulated will turn up
subtleties that might be ignored from a DSP standpoint. For example, many people
group the SSM 2040 lowpass filter and the Moog ladder filter together. It is
true that both filters work on the same basic principle (4 cascaded one-pole
filters with inverted feedback). However, each of them has entirely different
nonlinearites. The SSM2040, for example, has a very asymmetrical nonlinearity,
caused by the DC bias issues of the Darlington buffer (and the inability of this
buffer to swing through wide voltage values). Asymmetrical nonlinearities will
produce even-order harmonics, as well as the odd-order harmonics that are
commonly found in clipping. In addition, the input to each one-pole filter in
the SSM2040 is inverting - which means that a different side of the waveform
will be clipped for each stage. This results in a very complex nonlinear effect.
Reproducing even the simplest nonlinearities in the digital realm creates
problems with aliasing, numerical overflow, and so on, so emulating something as
subtle as the SSM2040 would be very difficult. Having higher sampling rates
would be very nice - at 96 khz, you won't have foldover of frequencies until
they reach 48 khz, and you won't be able to hear those until they fold back down
into the audio range (76 khz).
I doubt that any of the above has anything to do with Rasmus' original posting.
Oh well.
Sean Costello
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Josep M Comajuncosas wrote:
>
> I had a similar idea to emulate subtle variations of analog equipment. Maybe it
> would be nice to define a new set of variables, say ?i,?k,?a variables, which
> could have an added (and user definable) amount of thermal noise in them. The
> most important think should be the kind of randomness to be applied, and where.
It seems like at least part of this thermal noise could be emulated with randi.
The drift of analog VCOs, for example, could be controlled by this. However,
maybe you want to have a global temperature variable, which affects all ugens
that are controlled by the ?i, ?k, or ?a variables. Changing the temperature
would cause all of the associated ugens to change their behavior, but in an
unpredictable way (i.e. all the oscillators would drift upwards in pitch, but at
different rates and different amounts).
I'm not sure how you would emulate the "jitter" of analog VCOs, as I don't think
that this has been studied in detail. Some VCOs (Minimoog, VCS3) exhibit a
certain amount of pitch "jitter" that undoubtedly has an effect on the
perception of the sound. Perhaps noise through a bandpass filter of relatively
narrow bandwidth could simulate this effect.
I'd say that the tricky part of emulating analog equipment is emulating the
nonlinearities. An analysis of the circuit that is being emulated will turn up
subtleties that might be ignored from a DSP standpoint. For example, many people
group the SSM 2040 lowpass filter and the Moog ladder filter together. It is
true that both filters work on the same basic principle (4 cascaded one-pole
filters with inverted feedback). However, each of them has entirely different
nonlinearites. The SSM2040, for example, has a very asymmetrical nonlinearity,
caused by the DC bias issues of the Darlington buffer (and the inability of this
buffer to swing through wide voltage values). Asymmetrical nonlinearities will
produce even-order harmonics, as well as the odd-order harmonics that are
commonly found in clipping. In addition, the input to each one-pole filter in
the SSM2040 is inverting - which means that a different side of the waveform
will be clipped for each stage. This results in a very complex nonlinear effect.
Reproducing even the simplest nonlinearities in the digital realm creates
problems with aliasing, numerical overflow, and so on, so emulating something as
subtle as the SSM2040 would be very difficult. Having higher sampling rates
would be very nice - at 96 khz, you won't have foldover of frequencies until
they reach 48 khz, and you won't be able to hear those until they fold back down
into the audio range (76 khz).
I doubt that any of the above has anything to do with Rasmus' original posting.
Oh well.
Sean Costello
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From: Hans Mikelson
To: Csound list
Subject: Re: Yet another kick drum
Date: Sun, 29 Aug 1999 16:09:50 -0500
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Hi,
I think this one may sound a bit better than the last one.
Bye,
Hans Mikelson
;---------------------------------------------------------
; Kick Drum
; Coded by Hans Mikelson August 25, 1999
;---------------------------------------------------------
sr = 44100 ; Sample rate
kr = 44100 ; Kontrol rate
ksmps = 1 ; Samples/Kontrol period
nchnls = 2 ; Normal stereo
zakinit 50,50 ; May need this later
;---------------------------------------------------------
; Kick Drum 2
;---------------------------------------------------------
instr 11
idur = p3 ; Duration
iamp = p4 ; Amplitude
iacc = p5 ; Accent
irez = p6 ; Resonance
iod = p7 ; Overdrive
ilowf = p8 ; Low Frequency
kfenv linseg 1000*iacc, .02, 180, .04, 120, idur-.06, ilowf ; Freq
Envelope
kaenv expseg .1, .001, 1, .02, 1, .04, .7, idur-.062, .7 ; Amp Envelope
kdclck linseg 0, .002, 1, idur-.042, 1, .04, 0 ; Declick
asig rand 2 ; Random
number
aflt rezzy asig, kfenv, irez*40 ; Filter
aout1 = aflt*kaenv*3*iod/iacc ; Scale the sound
krms rms aout1, 1000 ; Limiter, get rms
klim table3 krms*.5, 5, 1 ; Get limiting value
aout = aout1*klim*iamp*kdclck/sqrt(iod)*1.3 ; Scale again and
ouput
outs aout, aout ; Output the sound
endin
f1 0 65536 10 1
f5 0 1024 -8 1 256 1 256 .5 128 .3 128 .1 256 .1
;a 0 0 5.68
; Sta Dur Amp Accent Q Overdrive LowFqc
i11 0.0 .18 30000 1 1 2 60
i11 0.5 . . . < 2.5 .
i11 1.0 . . . < 2 .
i11 1.5 . . . 1.5 3 .
i11 2.0 .18 30000 1 1 2 60
i11 2.5 . . . < 2.5 .
i11 3.0 . . . < 2 .
i11 3.5 . . . 1.5 3 .
i11 4.0 .18 30000 1 1 2 60
i11 4.5 . . . < 2.5 .
i11 5.0 . . . < 2 .
i11 5.5 . . . 1.5 3 .
i11 6.0 .18 30000 1 1 2 60
i11 6.5 . . . < 2.5 .
i11 7.0 . . . < 2 .
i11 7.5 . . . < 2 .
i11 7.75 .25 . . 1.5 3 .
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Date: Sun, 29 Aug 99 23:24:59 BST
From: jpff@maths.bath.ac.uk
Subject: Re: silly name restrictions was:Re: Phasor notes
To: csound@maths.ex.ac.uk
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Message written at 29 Aug 1999 16:09:19 +0100
Let me join in -- this name restriction is NOT silly. Let us remember
all the fuss over Russell Pinkston's cpoding of the Shepard tones,
which failed because a new opcode had captured a varaiabel name he
used. Since then there has been an attempt, not always successful, to
stop the introduction of opcode names which woudl break existing
pieces. I am responsible for the maligned gogobel. I am happy to
change it to bellagogo which is 'safe' if it is so much trouble. OF
course as upper case and lower case letters were separated spome years
back in Csound it is open to call a new opcode Allpassn but not
allpasn.
Quasimodo is entitled to correct what are now seen as design errors,
but we cannot do that for Csound. Think of the same with the Make
utility of Unix, which has serious design errors, but they are built
into programs all over the world.
=John ffitch
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From: Peter Neubacker
Date: Mon, 30 Aug 99 12:44:59 +0200
To: csound@maths.ex.ac.uk
Subject: Re: Insisting on PITCH
Reply-To: peter@orpheus.selene.cube.net
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I developed a new opcode for pitchtracking
based on the amdf method with an integrated
median filter to stabilize pitch output.
It works quite reliable on monophonic signals
and is quite fast - should work in realtime.
It is supposed to come with the next release,
you can find description and sourcecode at:
http://www.harmonik.de/femc/femcdown.html
Peter
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