I have been thinking about this problem and think that I've got a
suggestion now, but I'm struggling with the implementation. I'll
enclose the suggested patch anywayand hopefully someone will have
an idea how to fix it.
The equations I suggest for four signal amplitudes whose squares add
up
I made this mistake again last thursday - I computed sound amplitude of
some kind of omnidirectional source as being r^-2 where r is the distance.
However, I believe that it's energy that should be r^-2, therefore
amplitude should be r^-1. That's assuming ideal 3-D dispersion (no
ceilings and no
Hi, list,
I have practically abandoned the idea of making a subband adaptive
filter scheme using externals, but have not given up on making a
Newton-method, rather than a gradient descent method (Fourier
deconvolution is a Newton method, for instance, LMS and NLMS is a
gradient descent method).
forgot the patch...whoops
On 11/5/06, Charles Henry [EMAIL PROTECTED] wrote:
Hi, list,
I have practically abandoned the idea of making a subband adaptive
filter scheme using externals, but have not given up on making a
Newton-method, rather than a gradient descent method (Fourier
Thanks Martin. I will try it out and see!
Chuck
On 11/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I attach a version of lrshift~.c and the new help file. It now accepts floats
to change the shift amount. Maybe you could check to see if it works well. I
guess I could commit it to cvs if
The weirdness to which I was referring here was because rfft~
didin't have a nyquist component of the signal. So, the error was
around .001 for regular tones and varied according to slight phase
shifts, between 0 and 1 at the Nyquist frequency. So, I switched to
fft~ and ifft~ and the error
patch
idea1.pd
Description: Binary data
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I can't remember how to pass multiple arguments to an external. I
need to pass any number of float arguments. I think it goes something
like
void external_tilde_new(float *f_args, float num_args)
I've been searching thru the cvs repo to see if I can find an example,
but I haven't found one
Thanks, Frank and Mathieu
It is 100% crystal clear to me now. Thanks so much
Chuck
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How about something like this
[inlet~] [delread whatever]
| /
[+~]
| [bang~]
[/~ 1] [1 ] [+ 1] (the object 1 is a float with default value 1)
|
[tabwrite~ result] [delwrite whatever block_size_in_ms]
This is meant to be a recursive way of adding them up. You'll
wait a second
what are these signal blocks like that you want to average over? Is
it in fact a continuous signal, or are they actually independent
measurements of the same (phase-locked) data?
(If it's continuous data, you probably do not want this kind of approach)
Chuck
On 12/12/06,
I use the cc~ external. The time domain function is not so efficient
for large sizes, since it's an O(N^2) algorithm. The freq domain
version uses a fourier algorithm, so it uses only O(N*log(N)).
use [cc~ 1024] for time-domain where +/- 1024 is the bounds of the cc~
use [cc~ f] for one-block
What about efficiency? There may be certain advantages to defining
the data types, and constraining _inlets_ and atom types during
editing, rather than at run time. (that's just a guess)
Hm ... what do you want to say ? You want polymorphism ?
I say what I say. I'm asking, would we prefer
Maybe I'm missing the point here, but isn't Haskell a compiled (not
interpreted) language? We can add any compiled function, wrapped in a
C-written external, just so long as we have the symbols for the
function from the binary.
*OR* we can write an external in some funky language, so long as we
I think I understand now. This is quite a bit more complicated than I
had thought at first.
Have I missed the point entirely?
http://claudiusmaximus.goto10.org/gallery/coding/hsext/first-non-trivial.png
The middle window in the top row is the entire source for the object.
hsext provides
Check out vd~ (variable delay in milliseconds with interpolation) and
z~ (sample-wise delay from the zexy lib)
Chuck
On 1/22/07, Kim Taylor [EMAIL PROTECTED] wrote:
Hi everyone, I'm new to the list (and pd) so I hope this posts OK.
I'm creating a series of patches that rely pretty
On 1/25/07, Patco [EMAIL PROTECTED] wrote:
David Powers a écrit :
Hello everyone ...
SO, I finally got Linux installed on an old laptop (Debian stable
distribution).
Hello, what about using cvs, and trying to compile as much externals as
you could, I guess you will learn a lot through that.
Hi, Chuck
Biquad filters are based on a finite-difference equation:
y(n)=ff1*w(n)+ff2*w(n-1)+ff3*w(n-2)
w(n)=x(n)+fb1*w(n-1)+fb2*w(n-2)
Now, we can get rid of w, because it's basically a dummy variable and
write everything in terms of x and y. Y is the output, and X is the
input
I got it now. I don't understand why both expressions are useful, but
I see that they're the same.
Exactly. There's really no reason to use one or the other. The
two-equation form is the canonical representation, but the equations
can be written as a single equation, which sometimes appears
Nice patch. I like being able to put the poles and zeros somewhere
and then see the impulse response. That's cool. I would probably
change the pole radius param so that it gets asymptotically close to 1
without ever touching the unit circle (it took me a little while to
figure out how it
Hey, Kim,
I liked your patch, it has some very nice sound to it, and I like
being able to pan through the pluck/transducer location to hear all
the harmonics. That's really cool.
Now the way I see it, you could implement this as a sort of
ping-pong delay with the separate damping filters and
I have a feeling it's very well-known...one case is to have fH=2*fL,
which is the Morley wavelet...
On 2/3/07, Charles Henry [EMAIL PROTECTED] wrote:
There's a function I think could be useful for system response measurement:
f(x)=sin(pi*(fH-fL)*t)*cos(pi*(fH+fL)*t)/t
it's basically the ideal
There's a function I think could be useful for system response measurement:
f(x)=sin(pi*(fH-fL)*t)*cos(pi*(fH+fL)*t)/t
it's basically the ideal bandpass function, it's fourier transform is
0.5 between fL and fH and 0 everywhere else. There's another related
signal, the sinc... but what should
If you can (and it's a big if), try to suspend your acoustic dampening
panels above the heads of people. This will have a big effect on your
reverb problems without disrupting the space in the hallway.
I like the suggestions on this thread. very insightful
Chuck
Hi, list,
I have had trouble with pd-extended-0.38.4 lately. I once had it
installed, and it worked fine. Then, I upgraded to 0.39, and later I
decided to go back to 0.38. Since that time, I have had trouble
recognizing ALSA devices from the audio settings menu. Instead, I see
input device
wait a minute does that make any sense at all?
Post-re-install, I see no .pdsettings file at all for pd 0.38?
Was it all in my head?
Chuck
On 2/21/07, Charles Henry [EMAIL PROTECTED] wrote:
Thanks, Anders,
On linux, I just deleted .pdsettings in my home directory and ran
the installer
since it means that some settings
persist after deinstalling the 0.39 version.
/Anders
Charles Henry wrote:
Hi, list,
I have had trouble with pd-extended-0.38.4 lately. I once had it
installed, and it worked fine. Then, I upgraded to 0.39, and later I
decided to go back to 0.38. Since
When should you use rectangular notation and when should you use
polar?
Addition vs. Multiplication, of course
Addition must be done in rectangular coordinates
a+bi + c+di=(a+c) + (b+d)i
which we cannot do in polar coordinates, we have to convert back to rect. coords
Multiplication is
Hi, list,
I put together a few abstractions I wanted to share. These are just
basic operations for complex arithmetic (I left out conjugate,
addition, and subtraction, since I thought they were too basic).
cmult~
-- multiplies two complex signals; inlets are ordered from left to
right:
Hi, list,
I've made a drastic improvement over my previous room reverb patch,
that is, one that works. There's still plenty of room for changing it
up.
The patch calculates the room reverberation and saves it on several
tables on the first page, as freq. response, phase response, signal,
and
2007/3/6, Peter Worth [EMAIL PROTECTED]:
i kind of hoped that tabread4 would do some kind of interpolation
which might magically stop aliasing..
hmmm could we have some way of interpolating, just short of magic,
that would stop aliasing during downsampling?
tabread4~ takes an input of
more errors whoops!
sinc(t)=sin(pi*t)/t
sinc(t)=sin(pi*t)/(pi*t)
(2/N)*[1+sum{ j=1:N/2 ; cos(pi*t*fs*j) }
(2/N)*[0.5+sum{ j=1:N/2 ; cos(pi*t*fs*j) } ]
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The schematic of VOX CLONA (in attachment) gives you a precise purpose of
what I would like to
generate: one live voice (mine) generating an independant choir of 13 voices
with its independent
contrepoint projected in 3D sound space.
What's your concept for the counterpoint like? A canon
Hi, list,
I've finished analyzing the tabread4~ interpolation formula. It's a
real work of art, because it has a great low-pass characteristic, has
an efficient factorization, and has no phase shift.
I want to apply the tabread4~ scheme whenever the playback speed is
less than or equal to 1,
by the way, can anyone provide some insight as to how/why the
tabread4~ interpolation scheme was chosen in the first place?
(I have a pretty good notion from looking at Taylor series expansions
of G(w), but I'm still not sure what we would use for design criteria,
if we wanted to extend tabread4~)
Hi, list,
I've finished programming and (preliminarily) testing an external,
which is based on tabread4~. For normal speeds the operation is
identically tabread4~ , and for higher speeds, the interpolation
polynomial is modified to eliminate aliased frequencies. The
transition is smooth and
forgot to mention... march_of_the_sinusoids uses an abstraction
cnorm~.pd (attached)
Chuck
cnorm~.pd
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On 3/26/07, Derek Holzer [EMAIL PROTECTED] wrote:
I get nan when I blow up a filter or delay line. It's synonymous with
pure DC offset to me. The soundcard of course filters it, but it's
basically a pure positive or negative signal up to that point. Some
LADSPAs do it to me as well, like
I think there's an easier way to do directory recursion in makefiles,
just using the export command (which sets variables globally,
accessible to all shell scripts).
So, if you have a top-level makefile, which sets variables according
to a makefile.in, you can export those variables. Then, build
On 3/26/07, Martin Peach [EMAIL PROTECTED] wrote:
Does anyone know how to tell, in c, if you're getting nans? It should be
easy enough in the dsp routine to replace nans with zeros.
It's just a question of detecting them in time. I remember you could do
it in SANE, the old Apple math system,
okay, (on linux) you find math.h in /usr/include/ which includes a
file /usr/include/bits/mathcalls.h
there are three functions
int finite(double x) -- returns 1, if x is not NaN or inf
int isinf(double x) -- returns 1, if x is inf or -inf
int isnan(double x) -- returns 1, if x is NaN
int
On 3/27/07, Frank Barknecht [EMAIL PROTECTED] wrote:
Hallo,
padawan12 hat gesagt: // padawan12 wrote:
Amazing idea for a project Carl.
Yeah. Cool!
The way I see it you have two routes.
1) Do a full finite element physical model of a circular lamina
and measure the amplitude at
To me this whole idea seems rather difficult, if not impossible, to
implement. We are talking about interactions happening at the atomic
level of matter here.
It's not really the atomic level that we have to implement. You can
actually treat the material as continuous, and sampled at discrete
The patterns probably depend on the stiffness of the plate/membrane as
well as its shape, and the grain size and density of the sand.
It should depend on stiffness, density and shape. The speed of sound
in a material is sqrt(stiffness/density). The partial differential
equation for waves
filtering in general may not be the best approach because some of your
partials from one xylophone note will overlap with other note's
partials. They are inharmonic complex tones, which are not so easy to
predict you'll probably have to measure the frequencies of each
note of your xylophone
Robin - 1st order differential equation (specifies a constant phase
difference between incoming/reflected waves)
for accuracy sake the phase difference is dependent on frequency
(it works like an impedance)
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Hi, list,
I've created an abstraction for performing weighted median filtering
on a signal using 5 coefficients. It's a real cpu hog. It uses
heavily the expr, expr~ and fexpr~ externals, which leads to some
redundant calculations on each sample.
The test patch enclosed shows how effective
summarizing: Which is the next future of PureData
or How about this question: what does the roadmap look like for Pd v 1.0?
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I think it depends on the application for the most part, we can't
get a generic speedup from using multiple cores (forgive me if wrong)
that would apply to every single pd program. but some types of
computations such as large ffts can be performed faster when
distributed to different
I eventually got jack with pd running on my Debian machine. the Jack
package installs libjack version 100. I just made a symbolic link in
my /usr/lib directory with the command
ln -s libjack-0.100.0.so.0.0.23 libjack-0.80.0.so.0
(your specific version may be different)
Do you think this was a
I once wrote such a toolset that does automatically scale up
with multiple threads throughout the whole network. it worked
by detecting cycles in the graph and splits of the signals while
segmenting the graph in autonomous sequential parts and essentially
adding some smart and
Because it's cheaper to implement.
If well done, it's also an intermediate step towards automatic threading.
It's important to cut hard goals into easier goals, because it reduces
required investment and gives quicker returns.
I think that's a very good point. It could also lead to some new
I'm sure Mr. Spock had to deal with this all the time and I sure as
hell know that many people in my undergrad math department did!
~Kyle
my turn
[steps up on the soap box]
Everyone should learn at least calculus and differential equations.
Elementary concepts of signal processing (not
Mathieu's right here... It can be done with abstractions. I'm not
sure that mine are all correct, but I've used this technique a couple
times without making complete abstractions for it...
Here's an abstraction for computing a symmetric cross-covariance, this
way. It outputs the
Here's an abstraction for computing a symmetric cross-covariance, this
way. It outputs the cross-covariance, xcov(k)=sum(i=-32,...,31;
s1(i+k)*s2(i))
It's still pretty ugly, and the details of the math confuse me a bit.
I'm still working on the one-sided cross covariance function for
delays,
This one works within block sizes of 64, by using an [block~ 128 2] .
Now, I've got one where you add an argument and use arbitrary block
sizes. It's much more useful, this way.
[sxcov~ 2048] works within a blocksize of 2048, and calculates the
symmetric cross covariance of two signals.
Still
and
corrected. Hopefully, this method gives you the most flexibility for
computing sxcov~ and sxcorr~ in real time with abstractions.
Chuck
On 6/20/07, Charles Henry [EMAIL PROTECTED] wrote:
This one works within block sizes of 64, by using an [block~ 128 2] .
Now, I've got one where you add
It depends on the source, if its an impulse or if it's a periodic wave, for
which
correlation approach. I was reading this interesting one recently and
wondered if anyone tried it
Try using my sxcorr or sxcov patches from the previous thread about
cross correlation, with pique~ to find the
Another thought...
When you have the two microphones, just correlate the noise between
the two channels (not from the output/input1 and output/input2, but
between input1/input2). When the speaker is equidistant from each
microphone the delay is 0 (located at exactly half a block on the
output
.
Andy
On Fri, 25 May 2007 14:01:21 -0500
Charles Henry [EMAIL PROTECTED] wrote:
Hi, list,
I've created an abstraction for performing weighted median
filtering
on a signal using 5 coefficients. It's a real cpu hog. It uses
heavily the expr, expr~ and fexpr~ externals, which
On 8/3/07, chris clepper [EMAIL PROTECTED] wrote:
On 8/3/07, Steffen Leve Poulsen [EMAIL PROTECTED] wrote:
OK what about [walk] ?
[stagger]
[stumble]
[tipsy]
[blotto]
LOL
I never did care much for the term drunkard's walk. It's pretty old
fashioned. Also, not descriptive enough. We're
2007 11:51:24 -0500
Charles Henry [EMAIL PROTECTED] wrote:
On 8/3/07, chris clepper [EMAIL PROTECTED] wrote:
On 8/3/07, Steffen Leve Poulsen [EMAIL PROTECTED] wrote:
OK what about [walk] ?
[stagger]
[stumble]
[tipsy]
[blotto]
LOL
I never did care much for the term
At least it seems to be physically impossible to hit the hand, only a
finger could be cut off.
It doesn't seem like you could cut off a finger, with the blade
running lengthwise. No doubt it would hurt a lot, though. This is
all a bit morbid. Oh, well
I would trust Pd and the sensors in this
On 8/21/07, Tim Blechmann [EMAIL PROTECTED] wrote:
to make use of a multicore machine the only way to utilize all cores
is
to run several instances of pd, that are connected via jackdmp.
Now *there's* an idea. Would that really work? What would be the
downside -- aside from the
Something jumps out at me:
../libOSC/libOSC.a
this should be -L../libOSC -lOSC (that's a little-L, on the second one there)
I can't remember... what's the name of that command you use to list
the libraries that a binary links to? (I'm sure I've used it once
before, but not everyday :)
On
course it's a really handy tool. but since I can't look at an object
inside it's code, there was no way for me to know about it.
That is interesting to know. I didn't know that it was a standard
behavior either but since you've brought it up, now I'd like to
know how the code allows this
about pdpedia features:
Could there be a good way to cross-index (in pdpedia) with bug
reports, and show how they were fixed? Most user problems get posted
to pd-list, and any unresolved issues get placed in the reports.
A comment section is likely to be plagued with ambiguous user
problems and
How do you know it doesn't reflect on this list?
And more importantly, who cares if it does?
I think sexism is a subject that should be dealt with here on the
pd-list. And I think that the many replies here show that people on
the list do in fact care about it.
Allow me to try to condense the
As loing as it is not the crap beer we make in America! ;)
I know what you mean... I could go for a nice Stella Artois or a
Peroni... or a dozen Harboes...
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Since we're still talking about beers, I want to ask if there's any
pd'ers out there who homebrew? It is a favorite hobby of mine, though
I am not able to make a batch very often. I have on a couple
occaisions made an incredibly strong ale, not fit for human
consumption, including one stout that
on their own...
.hc
On Oct 11, 2007, at 3:20 PM, Charles Henry wrote:
Since we're still talking about beers, I want to ask if there's any
pd'ers out there who homebrew? It is a favorite hobby of mine, though
I am not able to make a batch very often. I have on a couple
occaisions made
there's no 'PLEASE STOP' there is 'YES PLEASE HURT ME!',
that's all we can take here.
That's funny... yes please hurt me is also my safe word
(just a little joke to illustrate how some 'fringe' vernacular enters
into common usage... we all know what that means)
but it is really kind of
what is this? I'm afraid to try
On 10/18/07, Enrique Franco [EMAIL PROTECTED] wrote:
http://www.hi5.com/register/jy2g3?inviteId=A_9b274a1_rSZw01uTU9p162690209
enrique
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On 10/18/07, Chuckk Hubbard [EMAIL PROTECTED] wrote:
On 10/18/07, Yves Degoyon [EMAIL PROTECTED] wrote:
ola,
this is an excellent summing up of all that is wrong here...
i didn't except to find soo much c.r.a.p. under the carpet ..
you can call me a troll, but i don't think i'm so
Any other ideas?
Another option is to use the 'plot as points' graph. You will get all
the points that way, even if the size is too small.
I'm a bit new to FFT in the pd context, but I think I grok Nyquist --
Sampling at S can, at best, yield the S/2 frequency (where S is the
sampling
That's cool, makes sense. Since I now understand that I'm dealing with
a graph/display issue, maybe I need to do some heavier lifting? That
is, unless somebody can suggest a better way, I guess I'll try and do
block-synchronized snapshots, somehow walk/traverse the fft results
myself and
On 10/22/07, Martin Peach [EMAIL PROTECTED] wrote:
Mathieu Bouchard wrote:
A very simple way to explain aliased frequencies would be: spin a bicycle
wheel. When you accelerate it beyond a certain point, it will begin to look
like it's going backwards instead. This is because the wheel speed,
On 10/22/07, Roman Haefeli [EMAIL PROTECTED] wrote:
On Mon, 2007-10-22 at 17:33 -0500, Charles Henry wrote:
On 10/22/07, Martin Peach [EMAIL PROTECTED] wrote:
Mathieu Bouchard wrote:
A very simple way to explain aliased frequencies would be: spin a bicycle
wheel. When you accelerate
That's not hard: just use expr~
[osc~ vibrato_rate]
|
[expr~ ($v10)*$f2*$v1+($v10)*$f3*$v1]
$f2 and $f3 are floats for your amplitudes for above and below respectively
Chuck
On Nov 6, 2007 1:00 PM, Libero Mureddu [EMAIL PROTECTED] wrote:
Hi list,
continuing my experiments with vibrato and
The curve won't have a continuous derivative, which may sound weird if the
curve is played slow enough. If ever your solution is not sufficient, it
might be better to try to come up with the integral of some continuous
function, but it may be a bit hard to make it align with the desired
I would suspect a memory leak. Your system will use up all its memory
for a while, then use up the virtual memory (swap space) which slows
things down by using the hard drive. Eventually it will use up all
memory and virtual memory, causing a lockup eventually.
ps aux | grep pd
can give you
On Nov 14, 2007 1:25 PM, Mathieu Bouchard [EMAIL PROTECTED] wrote:
On Tue, 13 Nov 2007, IOhannes m zmoelnig wrote:
no problem at all. [line~~] and [vline~~] will all come when the time is
ripe. and probably they will move from zexy into iem~~ :-)
[line~~] will come when the time becomes
What they are doing is increasing the accuracy of reading samples from
a large table, using 2 32-bit floats, instead of just one.
This [line~~] is a function of time, mapping time onto a 1-D path in the plane.
Tabread4~ works by pointer arithmetic. My guess what happens is, you
add the first
reprise, beat and such, are just larger scale splittings of the time
dimension in the same way that frequency separates from time. Reprises and
beats and rhythms are full of periodic patterns, just like the sound waves
themselves, but at a different scale, which doesn't make the physical ear
Personally, I like to use the sinc/rectangular impulse functions to
define the dirac delta, because it has a handy symmetry with fourier
analysis.
our rectangular function, in the time domain is
g(t)={1/(2T) , -TtT
0 , elswhere
which has fourier transform,
G(f)=sinc(f*T)
in the lim
On Nov 19, 2007 11:06 PM, Mathieu Bouchard [EMAIL PROTECTED] wrote:
On Fri, 16 Nov 2007, Charles Henry wrote:
I don't mean frequencies of sine waves, I mean frequency of any kind of
periodicity that is found.
Yes, I was sure you knew what you were talking about. I just had to
jump
I feel absolutely certain that I can convince you that timbre is *not* a
vector space, using only the defining properties of a vector space.
Ok, let's do that. How do you prove it?
With another little thought experiment. If I can't convince you, I'll
eat my words (yum)
First off, we need
On Nov 22, 2007 11:55 PM, Mathieu Bouchard [EMAIL PROTECTED] wrote:
On Tue, 20 Nov 2007, Charles Henry wrote:
Yes, but there is evidence for the fundamental bass that occurs between
pairs of notes, with a strength dependent on those ratios. Complex
harmonies could have multiple
On Nov 23, 2007 10:16 AM, Charles Henry [EMAIL PROTECTED] wrote:
I feel absolutely certain that I can convince you that timbre is *not* a
vector space, using only the defining properties of a vector space.
Ok, let's do that. How do you prove it?
With another little thought experiment
I remember there were many filter abstractions based on biquad~s. I
thought they were in the iemabs directory, but I can't find them.
With the biquad based filters, you can put them in a [block~ 1]
abstraction and modify the cutoff/center frequency at audio rate.
Chuck
On Nov 23, 2007 1:54 PM,
The problem with my examples, which I thought were bad was that
sometimes, I was using x(t) and y(t) as if they were signals, which
can be added and subtracted, and sometimes as vectors as functions of
time in an abstract timbre space.
Some of the presumed dimensions of timbre are things like
On Dec 2, 2007 11:52 PM, Charles Henry [EMAIL PROTECTED] wrote:
I would consider this function and its translations to be a convenient
basis for the set of continuous band-limited compact functions.
It is mainly useful because it allows this sampling property. If we
sample the function
On Dec 15, 2007 6:24 PM, Yvan Vander Sanden [EMAIL PROTECTED] wrote:
hi.
I am currently working on an external that generates rhythmic pulses in
a certain way. But I was wondering if I could run into problems with
calling usleep in an external. Alternatively, I suppose i could use a pd
timer
On Dec 17, 2007 3:03 AM, Frank Barknecht [EMAIL PROTECTED] wrote:
Hallo,
Charles Henry hat gesagt: // Charles Henry wrote:
[metro 1] creates a bang each millisecond, approximately. The message
rate is constrained by the block size, so you would want to put [metro
1] inside of a subpatch
On 12/19/07, Frank Barknecht [EMAIL PROTECTED] wrote:
I think, that now, that the negative numbers bug of [until] where
negative numbers acted like a bang, is fixed in the next Pd, maybe we
should tell beginners that they can send a number into Pd when they
are unsure if their patch is
On 12/19/07, Charles Henry [EMAIL PROTECTED] wrote:
I seem to be tuning in a little late, in this discussion, but if it's
a bad problem, couldn't you change the method of until to use only
floats?
oh, wait, now I get it. You should send a message back to until to
stop the loop at some point
On Dec 19, 2007 7:58 PM, Chris McCormick [EMAIL PROTECTED] wrote:
On Wed, Dec 19, 2007 at 02:22:44PM +, Andy Farnell wrote:
On Mon, 17 Dec 2007 22:23:11 +0100 IOhannes m zmoelnig [EMAIL PROTECTED]
wrote:
but a [bang(--[until] is not meant to loop infinitely.
it loops until a
On 12/20/07, Andy Farnell [EMAIL PROTECTED] wrote:
On Thu, 20 Dec 2007 10:43:57 -0600
Charles Henry [EMAIL PROTECTED] wrote:
I think a useful feature that would perhaps be able to handle this
type of problem is a 'halt'/'continue' routine for message processing.
Say, for example, it could
On 12/20/07, Mike McGonagle [EMAIL PROTECTED] wrote:
On Dec 20, 2007 10:41 AM, Russell Bryant [EMAIL PROTECTED] wrote:
So, after going through my own mental exercise to analyze the situation, I
now
don't think any changes should be made at all.
I agree with this. This is just one of
-0600
Charles Henry [EMAIL PROTECTED] wrote:
On Dec 19, 2007 7:58 PM, Chris McCormick [EMAIL PROTECTED] wrote:
On Wed, Dec 19, 2007 at 02:22:44PM +, Andy Farnell wrote:
On Mon, 17 Dec 2007 22:23:11 +0100 IOhannes m zmoelnig [EMAIL
PROTECTED] wrote:
but a [bang(--[until
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