Frank Barknecht wrote:
difference between both is illustrated in attached patch.
thanks, ok, i think i will use metro in most future
pd-sequencer patches, instead of my old phasor-method...
it was a 'quick-and-dirty' solution anyway, since i have sampled
the phasor with metro 3 or 2 and
threen wrote:
im not defending/attacking anything ... its just the fact that the
analog cliped signal sounds much more convincing when comparing with
digital hysteria
hmm, yes, maybe. i think it depends on the context where the sound is
used...
btw: are there any good saturation/distortion
James wrote:
i'd like to take an informal survey. is it true that you can do no serious
work with puredata using the jack audio server for linux and mac os x?
no, i have no major problems with jack (jackd 0.100.0) pd (0.39-2)
on linux (ubuntu dapper), using a m-audio 2496 soundcard.
though i
please apologize if this is not the right list to report
problems with pd-extended.
i have just tried pd extended on windows, and encountered the
following problems:
-iem abs vcf_hp2,4,6,8 are missing (i copied these from my linux
installation)
-iem_cot missing, used by vcf_hp2 and vcf_lp2 (i
Hans-Christoph Steiner wrote:
-the crashes when closing complex patches seem to happen
more often compared to the linux version.
If you can narrow this down, please post a bug report, hopefully with a
example patch, to the bug tracker:
maybe it is just not true that the 'crash on
Steffen wrote:
Isn't this much like what you/we talked about late november last year
wrt. your groovbox patch? This rings a bell that's why, please check if
it's related to this bug report, which, i think, boils your problem from
last year down to a simple example:
a while ago i have built a little, 'quick and dumb' beatloop
which uses not exactly the same techniques used in live (it is
missing the tempo-warp-stuff), but does basicly the same thing as
reaktor and acid do. without any analysis though, so
transients are not taken into acount. i have built
)+res;
#X text 39 220 cosine;
#X text 147 191 sine;
#X text -310 363 multiply input by attenuation factor;
#X text -404 106 4/2008 martin brinkmann;
#X text -405 8 slightly crude (especially the attenuatiun factor) lowpass
filter design \, made without knowing too much of the math involved.
frequency
Isidro Gonzalez wrote:
so is it possible to get the gain from the poles and
zeros, without
calculating all coefficients?
Of Course it is. You must evaluate the transfer
function of the filter in the unit circle.
For a nice explanation on how to do this and some C
code samples, see:
hi list!
i have a problem with iem-lib (r 1.15 (the most recent one, i hope))
splitfilename: names wich are exactly 76 characters long cause pd to
freeze (see attached example). is this a known/fixed bug?
i could not find anything in the archive or on the net.
and is there a workaround (or an
IOhannes m zmoelnig wrote:
coud you (re)attach the example?
oops, sorry! i have clicked send a little too early...
you can use zexy's [symbol2list] for such things as well.
thanks, i will take a look at it.
bis denn!
martin
#N canvas 481 272 608 289 10;
#X obj 15 217 splitfilename;
Hans-Christoph Steiner wrote:
That's a new one for me, sounds pretty crazy.
it was related to my outdated (1.15) iem-lib binary. i have
just tested with current pd extended version, and copied the
latest (r 1.17) iem-lib ito my current pd installation, and
the problem (quite crazy indeed) is
Steffen Juul wrote:
It doesn't freeze in Pd-extended 0.39.3 (mac osx). No idea what version
of iem-lib that is though.
it seems that it only happend with my old iem-lib version. installing
the latest one (1.17) solved the problem.
bis denn!
martin
Ctrl Alt Back wrote:
Works ok on debian Pd version 0.41.0-extended-20071107.
works with recent version of the iem-lib, it seems that it only happend
with old iem-lib versions, maybe only with old iem-lib + new pd version.
thanks for checking to everyone!
sorry for any inconvenience.
bis denn!
hi!
i'm sorry if this has been asked (and answered) before,
but i have not found a solution yet. (in the list archive)
~
~
(and maybe other signal operators from zexy)
are not created. is there a workaround?
vcf_hp2~ throws an iem_cot~... couldn't create,
but only if not vcf_hp4~ (or some other
Si Mills wrote:
I've been using [expr~ ] as a workaround
eg. [expr~ $v10.5]
ok, it is good to know that there is a workaround,
and i will probably use the expr-method in the future
for compatibility with 'pd vanilla'. but i am not
looking forward to converting all my patches...
thanks for
Damian Stewart wrote:
to answer my own question:
use [biquad~] with coefficients taken from the [lowpass] external.
i while ago i have tried to build a resonant lp with biquad~, but had
problems with clicks, when the coefficient were changed (even with quite
slow changes). only very slow
Frank Barknecht wrote:
that is true, but it looks like no one has made a (usual lp,hp,etc.)
filter with these objects until now.
Except Miller. [1]
you are right, there are some usual filters made with
pole/zero objects, the shelving and peaking ones, a nonresonant lp
and a bp, iirr. and the
Damian Stewart wrote:
[shrug] i don't even know what a 'zero' is.
a (probably a little too simple) explanation:
a zero is a valley. its depth is 0 on its deepest
point. a pole is a very (very,very,...) high
mountain. both things exist on the complex plane
(just a plain, where all points have 2
Ben Baker-Smith wrote:
-Does the netbook have enough processing power for general GEM
applications? I'm usually not dealing with video files, but rather particle
generation, shape manipulation, GIF texturing, and audio-response. On my
Macbook (2.0 GHz intel processor) the CPU meter always
Solen Music wrote:
I did mean step sequencer.
i have allways been interested in step-sequencers, and made
a few attempts to build a good one.
of course i was not successfull in making the perfect
step-sequencer... and i would not reccomend my creations for learning,
since they are quite messy,
Solen Music wrote:
my plan for the tappable groove metro is not to use [metro] but to
keep a variable table of times between output bangs. i hope to hook
two drum pads to a [timer] based kind of bpm counter. one pad will
input the downbeat times and the other the offbeat times (the usual
Derek Holzer wrote:
VANILLA PD
*SIGNAL CONTROLLED*
vcf~
moog~
is moog~ really in vanilla? (not here (0.42-5), unforunately, though i
think it should be...)
bis denn!
martin
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brandon zeeb wrote:
Can you reproduce this behavior on your machine?
no, both filters sound exactly the same here. with 44100
samplerate though.
bis denn!
martin
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James Dunn wrote:
Can anyone help?
probably not. i have had the same problems since i upgraded from hardy.
i had more or less periodic clicks/dropouts and erros, especially whith
heavy gui activity. (minimizing the gui while recording helped a little)
i have given up, and use oss instead of
Otto Maddox wrote:
Maybe it's just because I'm new to the world of mobile computing and
have missed something, but I couldn't find a single mobile
hardware/software platform which is properly Free/open from top to
bottom (i.e., friendly to people who don't mind getting their hands
dirty),
Hans-Christoph Steiner wrote:
as far as i have heard, rjdj is currently ported to android.
But not released, and likely not free.
probably free as in beer only, but i hope it does not take too
long until it is released.
What's non-free about Maemo? I use a Maemo device daily and as far as
Hans-Christoph Steiner wrote:
as far as i have heard, rjdj is currently ported to android.
But not released, and likely not free.
probably free as in beer only, but i hope it does not take too
long until it is released.
AFAIK, they wanted to keep the core rjdj app a free download, but
sorry, this is probably not very helpfull, but:
Matthew Logan wrote:
Has anyone had any issues with installing gp2xPd? I can't get this thing to
work.
it is runnig fine here on exactly the same device (f100b, 3.0.0)
and i vaguely remember that i had a few issues with the installation,
but
Matteo Sisti Sette wrote:
I write just to see if anybody has already seen this happaning
happens here also sometimes. closing and reopening pd
usually solves the problem. (ubuntu jaunty 32 bit,
pd .42-5, gem loaded, but i think it happend also without gem)
and can
give me some clue as how to
David Schaffer wrote:
Idealy the device should be fat sounding and resonant...
imho the nusmuk-audio-filters posted to the list a while ago
are among the best sounding (pd vanilla) filters. only the
cpu-load is very high, due to fexpr~.
bis denn!
martin
colet.patr...@free.fr wrote:
didn't like the sonority of netpd for example, because rendered
texture are poor, only one sytnh sound good
the reason might be that netpd is 'pd-vanilla', and there are
not so much techniques used for getting smooth(er) sound (afair),
like bandlimited oscs and
martin.pe...@sympatico.ca wrote:
If a basic [osc~] sounds like 8 bits I think maybe you need a better sound
card. ;)
i did not mean that literally ;) and i like 'lo-fi-sound'.
bis denn!
martin
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Roman Haefeli wrote:
Actually, there vanilla-based bandlimited oscillators and some
netpd-synths are using them.
maybe the one good-sounding synth is one of them.
and i remember some synths which use just phasor~-0.5
(which i also use quite often).
bis denn!
martin
Roman Haefeli wrote:
you can find the major7.pd patch and its depencencies here:
http://www.romanhaefeli.net/software/pd/
i could not find major7.pd there, but i have tried the
bandlimited_oscilators (sinesum), which sounded quite a lot
better/smoother than other methods for making
bandlimited
Jerome Covington wrote:
Are there any recommendations on netbook/audio interface combinations
for further development and performance using pd?
i would recommend not to use a netbook, or any other atom-based system,
because the performance with pd is rather poor. my aspire one with
1.6ghz does
i often use sounds from the olpc project:
http://wiki.laptop.org/go/Sound_samples
there are also a few instrument-multisamples.
and here are quite a lot samples of piano and
orchester instruments:
http://theremin.music.uiowa.edu/MIS.html
bis denn!
martin
Alex Lucas wrote:
http://en.flossmanuals.net/PureData/GeneratingWaveforms Any help with this
would be very much appreciated, it may just be my soundcard
acting strangely.
i think the artifacts (audible at higher frequencys) resulting
from the phasor going from 0 to 2051. after changing this
Alex Lucas wrote:
there doesn't seem to be
any visible clipping on the table. The audible clipping was present across
the frequency range but changing the multiplication on the phasor~ to 2048
has solved the problem.
it is not so much clipping, but a discontinuity in the waveform, which
David Schaffer wrote:
I'd like to know if there's a bit crusher/ lo-fi/ sonic destructor abs.
a bit-reduction effect is quite easy to build:
multiply the signal by a (larger) number,
throw away the fractional part (by using expr~ int($v1), or building
something with wrap~)
and divide the
the ~ and ~ etc. are not created, like in the previous versions.
(linux/mac/win).
this is id 1702883 in the bugtracker, but it happens on all os,
not only osx.
on windows i had a errormessage allocting font resource or something
like that (sorry, i can not remember the exakt message), during the
On 06/29/2010 02:20 AM, João Pais wrote:
Basically
something that allows people to disseminate the program (even though is
available online), but without them changing it (or not without my
consent/knowledge).
licence-wise you could use some properitary
free(as in beer)-ware licence which is
On 06/30/2010 06:12 PM, Mathieu Bouchard wrote:
imho this would not be worth the effort, and a horrible idea to make
deliberatly unreadable pd-patches...
If you keep on saying things like that, I will make a patch obfuscator
patch just so that the content of threads about obfuscation would
On 08/17/2010 09:52 AM, hghoyer wrote:
I have the FM patch improved a little. With the link it can be downloaded. I
welcome feedback on the patch. Maybe someone else has a suggestion for
improvement.
i think i would add a slider (or better number2) for the
frequency ratios, which are fixed
On 08/16/2010 06:21 PM, Pierre Massat wrote:
Hi,
I've been thinking of writing a pd version of autotune, but before i get
started i m wondering if anybody as ever tried doing this?
not in pd, but in reaktor and c. combining pitch-detection and a
pitchshifter (and some quantization) works
On 11/28/2010 09:21 PM, Andrew Faraday wrote:
also, I can't get this to work on my mac, not entirely sure why, oh well.
never mind.
works here on macos in pd extended 0.42.5. the cpu load is a bit heavy
though, on my 2009 mac mini. (about 70 percent) i also got a few
r13 couldn't create.
on
On 11/29/2010 11:09 AM, Andrew Faraday wrote:
How did you use the data from your grid to generate the music?
my grid is a (virtual)tenori-on-ish sequencer, and:
Was it the straight-forward each position represents a note approach,
exactly.
bis denn!
martin
On 02/07/2011 11:21 PM, Pedro Oliveira wrote:
Last weekend I did a small performance at the Hochschultage at the HfK
Bremen, Germany. This is from a project called Performance for One which
is a small part of my research - more info about that soon ;)
unfortunately i missed your performance
On 02/09/2011 01:02 PM, Pedro Oliveira wrote:
But about the technical setup it was my first time ever performing live with
Pd, so I kind of created an ensemble of possibilities I could work with,
however limiting myself to only four channels...
i think it depends on what is going on in a
On 03/25/2011 09:44 PM, Xavier Miller wrote:
Wouldn't be possible to mix the best of both worlds (tabread4~ and
sfread~) by pre-loading the first (milli-)seconds of each sound in
memory, then switch to streaming
i remember a couple of years ago something like this was posted to the
list.
On 04/21/2011 01:48 PM, Joe White wrote:
http://kemptonmooney.com/2010/09/pure-data-midi-note-off-solution/
this does not happen here, using the method from the monosynth example
in the pd-help. (stripnote), 3. c10. however the sound stops, if the
last pressed key is released. if this is not
On 05/26/2011 01:29 AM, Martin Dupras wrote:
In other words: let's say you want 5 beats in a grid of 12 (or a
12-sided polygon, if we use his graphical representation), the exact
spacing between two beats would be 12/5, or 2.4.
The first beat would be 0*2.4= 0.
The second beat would 1*2.4
On 06/20/2011 01:43 PM, Marco Donnarumma wrote:
Ingo,
thanks for your explanation, I think to understand how he's playing.
i do not think i do (completely), though i have only watched it twice
so far. it seems that everything is based heavily on live-looping. at
first i thought he was using
On 06/21/2011 08:05 AM, Ingo wrote:
All lines are played live. There is no pre-recorded drum loop. He is using
audio looping only. The drums are played note by note.
...
you are right, and after having watched the video another time,
i agree to your observations. (and meanwhile it has been
On 08/23/2011 11:47 PM, Richie Cyngler wrote:
(whatever kind of sound gen you like)
|
[*~ 0.2]
| \
[dac~]
i think you should put a clip -1 1 before the *~ 0.2.
otherwise, if you connect something like osc~ *~ 1 for
example, you would get some very loud drilling sound at the
dac~
On 09/01/2011 11:51 AM, Rick T wrote:
For more background I'm going to have the Frequency, Amplitude, and Phase
controlled by Midi Controllers. So there will be a total of 9 midi
controllers used (3 for each signal).
i think independent control of the 3 frequencies makes this a little
On 09/04/2011 01:46 PM, martin brinkmann wrote:
and reblock the patch to
fit the frequency of d. this would always guarantee exactly appended
waves, though not with precise control over the frequencies of b and c.
...and of course somehow write the result to normal (power of 2)
sized blocks
these minimal patches are fun! the idea reminds me a little of the sc140
project:
http://supercollider.sourceforge.net/sc140/
here is my 1st one.
bis denn!
martin
#N canvas 424 308 387 350 10;
#X obj 125 242 dac~;
#X obj 48 227 cos~;
#X obj 90 133 *~;
#X obj 52 53 phasor~ 1;
#X obj 131
and another one,
bis denn!
martin
#N canvas 559 407 450 300 10;
#X obj 83 251 dac~;
#X obj 76 203 osc~;
#X obj 141 68 wrap~;
#X obj 121 127 vd~ \$0delay;
#X obj 131 234 delwrite~ \$0delay 1000;
#X obj 75 181 mtof~;
#X obj 68 33 phasor~ 0.707106;
#X obj 259 31 phasor~ 0.866025;
#X obj 75
improved version of my 1st one (got rid of the lame random), another
feedback-based, and a fm-tuningfork with echo.
bis denn!
martin
#N canvas 586 335 387 350 10;
#X obj 160 252 dac~;
#N canvas 826 407 450 300 stuff 1;
#X obj 48 227 cos~;
#X obj 90 133 *~;
#X obj 52 53 phasor~ 1;
#X obj
after i had fun making very small patches which make nice noise,
i got the feeling that i was slightly missing the point:
On 10/07/2011 02:25 PM, Chris McCormick wrote:
http://countercomplex.blogspot.com/2011/10/algorithmic-symphonies-from-one-line-of.html
when i had actually read this.
On 10/16/2011 02:31 PM, hardoff goes bananas wrote:
further condensing of martin's patch:
that is probably the most compact version (of course only if expr counts
as one object...) it sounds a little different from the message based
version though, but this is probably only a matter of tweaking
On 10/18/2011 03:36 AM, hardoff goes bananas wrote:
the main dilemma here is that the patch runs at normal samplerate and
bitrate.
lower samplerates are possible with smaller numbers as input for the
counter, and rounding to int
i think to get the sound close to the original code examples,
On 10/19/2011 07:56 AM, Mathieu Bouchard wrote:
Something like ((y^128)-128)/128.
and some output filtering to eliminate frequencies above 1/2 of 8 khz.
i have just added a few lop~, and i think it sounds quite close now.
bis denn!
martin
#N canvas 101 287 793 443 10;
#X obj -61
On 10/22/2011 02:39 PM, saskia diez wrote:
I just discover this since i bought a nice speakers, before i
havent realize that sound editors add that hiss.
i could hardly believe that, but you are right! i have just created a
file with writesf~, ensured that it contained only zeros (hexeditor),
On 10/22/2011 03:49 PM, Pierre-Olivier Boulant wrote:
I think you are talking about this:
http://en.wikipedia.org/wiki/Dither
http://en.wikipedia.org/wiki/Noise_shaping
yes, and it is possible to disable this in audacity
(edit/preferences/quality setting dither to none)
bis denn!
On 10/23/2011 08:20 PM, Mathieu Bouchard wrote:
How much is a lot ? What's the amplitude in the noise ? If it's
something like 1/65536 of the maximum level, that's not what I'd call
« a lot »...
it is more. after normalizing i had about 8 different
sample values (including zero) in the file.
On 02/04/2013 12:49 AM, Esteban Viveros wrote:
Hello,
Have pd-extended .43.4 a equivalent to max/msp rslider object?
you can rather easily build something similar in vanilla
using datastructures. a quick (maybe a little clumsy)
example is attached. it is not exactly the same
(extend-mode
On 03/29/2013 01:10 PM, matijoncek prdoncek wrote:
I was wondering if it's possible to change the the number of pixles by
which the objects move when you select them and use shift and the cursor
buttons to move them.
it is possible using the displace msg and pointer from the struct
object,
On 04/17/2013 10:00 PM, Thomas Mayer wrote:
is there a 2D slider object, similar to a combination of a
hslider and vslider, something like the Kaoss Pad touchscreen?
thanks to both of you, but both objects are not quite what I had in
mind.
[grid] behaves exactly like the kaosspad surface.
On 05/28/2013 12:11 PM, Roman Haefeli wrote:
Hi all
Is it possible to have a scalar with a settable position, but which
cannot be moved with mouse interaction?
yes, by adding a -x to the drawpolygon in the template.
I figured I can use arrays to create grids of immutable scalars, but
then
On 08/19/2013 08:11 AM, Miller Puckette wrote:
Hi all -
I've now put out test 2
there seems to be a new behaviour concerning tabread4~ and upsampling:
(compared to 044, i have not checked the 1st 045 version)
in 044 the frequency of the phasor reading the table had to be
multiplied by the
On 08/27/2013 05:41 AM, Pagano, Patrick wrote:
I have a metro on about 200 triggering about 9 bangs for a [random 127] and
the values are bopping all over the place and i would like them to move
smoothly from random number to random number like a knob would. Can someone
help me out with
On 10/11/2013 08:32 AM, Dan Wilcox wrote:
[vcf~] sounds close, but of course, it's a band pass so it's not a real
replacement.
the undocumented 2nd output of vcf~ is a lowpass. and there are a few
methods to build the basic (cookbook) filters with pd-vanilla-objects:
you could use fexpr~ to
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