yeah cheers. may be useful for me with a project i am working on.
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How is it possible to change in real-time the start and stop position of a
soundfile in loop mode without getting audible audio clicks.
other options are:
(1) use 2 loop players and crossfade between then every time you change the
position. (you can put each player in a separate subpatch
really i know close to nothing about midi handling in pd, but i remember
reading somewhere that midi latency is connected to the audio latency
because midi information is passed along with the audio. so, perhaps that
has something to do with the message-length restriction?
synth secrets has a few chapters on brass instruments:
http://www.soundonsound.com/search?page=2Keyword=synth%20secretsYear=%20Month=%20Words=AllSummary=YesSection=0Subject=ShowResults=yes
starting at chapter 24.
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and the didgeridoo, apparently. if i made a mistake as to what a brass
instrument actually is, then fair enough. it's confusing.
http://en.wikipedia.org/wiki/Brass_instrument#Some_other_brass_instruments
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spot the difference:
.tar
.rar
did you try just double clicking on the icon? been a while since i used
mac, but i think installing is really that simple now.
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What I'd like to see is interfaces like Live implemented in Pd. Pd already
has all of the audio tools, what it is missing is the GUI toolkit. One of
my goals with pd-devel and tkwidgets is first, to make some good widgets,
and second, to make it easier for people to code intricate GUI
would widgets and whatnots be subject to the same clunky slowness of other
pd gui objects? or is this something that might be improved too?
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Luke, i played with that briefly yesterday,
when closing, i got a whole bunch of discard changes to this window?
dialogues.
(also, slightly off topic, but i think it was you who mentioned a while back
that you were working on matrix-editor sequencer in DS. is there any
progress on that? )
just watched the vimeo clips. fantastic!
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i haven't tried this new version of pd yet, but i know that if i did, i
would agree with the comments above.
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frank, you should post these things up on the pd forum too.
or if that's a hassle, do you mind if i relay them on for you?
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that is one of the best looking pd interfaces ever. really inspiring.
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this seems to be missing the rc-square~_ abstraction.
...and the rc-sssad stuff.
anyway, looks great dan. cheers.
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yeah, i just add -- as an argument.
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i have some GOP abstractions which use the sssad system for state saving, so
they need a creation argument to set their 'key' for sssad to identify
them. as so often happens, i load up a bunch of these abstractions and
start playing around with their sliders until it sounds good, and i decide
to
hi frank, thanks for the reply.
your system uses extra inlets and outlets on each abstraction to load the
preset data, which is something i wanted to avoid if at all possible. but
saying that, i think your system has one really great advantage which i
didn't imagine by myself... the preset data
i just added a pd patching idea, to build a library of filters.
i think this idea would also work really well if merged with the 'blosc'
one, because frankly $4500 for a student to make some bandlimited
oscillators is pretty expensive!
http://puredata.info/dev/summer-of-code/PurePDFilters
a very simple solution would be to use a subpatch instead of an abstraction
in the help files.
[pd output~]
this would also remedy the problem reported more often: why doesn't the
output~ object work when i copy the help files to my desktop?
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* create a subpatch
* create an object box in the subpatch, but don't type anything inside the
box
* go back to the parent patch and convert the subpatch into another object
(i tried to make it a [send])
* enjoy pd freeze
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a compressor?
there's a good one in the net-pd library.
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i think this idea of making a unified library of patches is fantastic. but
i don't think it should be a project given to a student with little
experience in pd.
i'd be more than happy to help out.
what would be the best way to set up communications between us, if we take
this on? that would be
is there a way to read .syx files in pd? these are in raw hex format.
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to re-create the sound of a dx7 i need to change samplerate from my default
(44.1khz) to 28khz.
globally changing pd's samplerate to 28khz has the intended effect, but i
want to be able to run other patches at 44.1khz while running my dx7 at
28khz.
i naively tried adding this construct to the
so, i thought about this more, and realized that 28hz is quite close to
being 2/3 of 44.1
so, it MIGHT be enough to do the following:
for every 3 samples, output the first sample, the second sample, and then
instead of playing the 3rd sample, repeat the second sample.
so, if my block looks
But I think that we don't really know well enough how to do the whole
thing right
i have good experience in doing a library wrong. :) i can at least point
out the flaws in that so they don't get repeated.
none of the libraries are very good general all-purpose toolkits though, in
my opinion.
hi martin, thanks for taking the time to reply.
i can set pd to whatever samplerate i want in media-audio settings. i just
checked again going through a whole bunch of values between 22050 and 44100
and they all work.
my problem is that the samplerate is global, and i need it only to be
local.
it's not possible to filter without ANY roll-off as far as i know.
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chebychev filters look good, elliptic ones look better:
http://en.wikipedia.org/wiki/File:Electronic_linear_filters.svg
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actually, i just checked and that method i suggested before doesn't work at
all. it's the same as the [phasor~ 28000] - [samphold~] effect.
martin, i missed this part of your mail before:
Unless you're looking for aliasing artifacts, wouldn't it be easier to use
a multi-pole low-pass filter
yeah sorry frank, i should have explained more clearly.
i also think that no GUI is the way to go for functional abstractions. that
was the big flaw of the DIY library i did, that the function of the
abstractions was tied in with the gui component. i did it that way because
i didn't want to
One thing that occurs to me is that the quality of the reconstructed
sound has more to do with the interpolator rather than the sample
rate.
i have a feeling that because the synth has non-bandlimited oscillators,
that the sample rate itself is pretty important, or more importantly, the
aliasing
so if those aliasing effects are going to help, then maybe a big step will
be to ADD aliasing. there are a couple of suggestions here:
http://www.music.columbia.edu/pipermail/music-dsp/2002-March/047953.html
and here:
http://www.music.columbia.edu/pipermail/music-dsp/2002-March/047966.html
i had a minor breakthrough.
upsampling my signal path, by using [block~ 64 1 2] has the same effect on
the sound as lowering pd's global sample rate to 22050.
not perfect, and to me it defies logic, but certainly a lot closer to the
28khz sound than the 44.1khz sound was getting.
YES!
thanks hans, and thanks mr peach!
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make a normal counter:
[f ]x[+ 1]
with the outlet of [f ] going to the LEFT inlet of [+ 1] , and the outlet of
[+ 1] going to the RIGHT inlet of [f ]
then,
[wrap -99 100]
|
[abs]
if your limit was 1000, you would do [wrap -999 1000] ..etc
hope that works.
so are we going to go ahead with this?
does anyone want to volunteer to set up a wiki? i'm terrible at any of that
sort of thing.
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if it is a problem to use [wrap], this should also work:
[bang(
|
[f 100]x[+ 1]
|
[mod 200]
|
[- 100]
|
[abs]
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there are a few EQ units in my DIY library:
http://puredata.hurleur.com/sujet-1982-diy2-effects-sample-players-synths-sound-synthesis
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cheers. very useful!
silly question probably, but is biquad~ 100% vanilla? or is it sort of
'caramel' like expr ??
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please please please,
make one with:
[bang(
|
[until]
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that's kind of cool, because it makes absolutely no sense when you look up
close.
On Wed, Apr 1, 2009 at 6:21 PM, Max abonneme...@revolwear.com wrote:
make this into a shirt:
Am 01.04.2009 um 09:05 schrieb Derek Holzer:
If you made one like this:
thanks hans,
i was waiting to play with this.
cheers.
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yeah! that cake is awesome!!!
how did it taste?
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yeah weird. same thing here.
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i just installed pd-0.42-4, but i can't get midi working. pd -listdev says
i have no midi devices.
The alsa Aconnectgui interface doesn't list pd as a midi device either, so
my guess is that i missed something while installing.
the install.txt file says to add this flag:
--enable-alsa
so, i
YES! thank you Ed - you're a mega legend!!!
pd is now controlling the squelch of my MKS-50 alpha juno rack.
beautiful.
.matt.
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a really good way to practically demonstrate PD is to build a simple patch
in front of people. maybe practice building that patch a couple of times
before you do the presentation.
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hmm, i don't know. are you using a recent version of pd?
i have dynamically created audio abstractions with no problem.
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what about something like sending a value to the inlet of an audio
arithmetic object? would that update the dsptree?
i'm wondering why the audio in my abstractions is working, because i don't
think i am doing any of those things.
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hi steve, welcome!
frank forgot to mention his excellent spectral delay.
http://footils.org/cms/weblog/2007/feb/20/beginners-guide-fft-objects-pd/
also search for the following for a bit of fun:
netpd , rjlib , diy2
just google pure data plus those words and yoU'll find them all.
heaps more
just a sidenote, but you do know about [switch~] to turn off audio
processing in a subpatch/abstraction, right?
i used to have a patch that had about 3000 dsp building blocks (oscillators,
filters, waveshapers, eq, effects, etc), and the patch just switched on each
part when it was called. in
is there any way to make the 'open' dialogue window ignore hidden
files/folders? this seems to happen automatically with the version of
pd-extended i have.
(linux, ubuntu)
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ok, cheers, i'll search it out.
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i find that you can do a lot by building a simple arpeggiator sequence of
just 3 or 4 notes, and then send messages to do things like: retrigger,
reverse, transpose, change speed, etc
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running mac os 10.4, and the latest version of vanilla pd with -nogui, the
attached patch does not work how it should.
the patch is a simple phasor~ ping sound, run through a [delread~] /
[delwrite~] delayline. with gui turned on, it works as expected, making a
feedback echo,
but with -the
yeah sorry, i tested on 0.42-5, and also on 0.41-4 and got the same thing.
your help on this issue is really appreciated iohannes, thanks.
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would it be possible for us to modify the source code (eg d_delay.c) to set
a non-variable samplerate? in our case, the samplerate will always be
22050hz.
or is it not as simple as i imagine?
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i posted a realtime granulator on the pd forum a while back. you may need
to log in to the forum to get the attachment.
http://puredata.hurleur.com/sujet-1932-grannie-basher-delay-line-granulator
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I made a patch which times the duration between two bangs sent by [bang~].
If i set the blocksize to the default of 64, then each bang comes 1.4ms
apart - as expected with 44.1khz samplerate.
However, making the blocksize smaller has no effect on the duration between
the bangs.
Does bang~ have a
i wanted to send a bang on a zero-crossing
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i wanted to make some effects which acted on segments of sound cut at zero
crossings.
i guess i will just try them out on tables, like in your patch above.
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i'm still at a loss to explain the effects observed in the attached patch:
blocksizetester2.pd
Description: application/extension-pd
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wow, monumental news!! big toast!!
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[tabindex~] is the best one so far in my opinion.
not sure if it'd be technically correct, but what about:
[vtabwrite~]
???
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i had some problem with a -nogui patch running [delwrite~],[delread~]
pairs. it was a C programmer who eventually fixed the problem for me, but
if i recall correctly, the issue was with the delayline samplerate not being
set properly when pd is running nogui. running with gui seemed to introduce
oh, and the bad news was that the only way we could fix that problem was by
modifying the pd source code.
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once again, here is my abstraction that does everything exactly as the
[spigot~] object does.
if someone would like to include this in the 'extra' folder for pd-extended,
it may be helpful to a few people.
spigot%7E.pd
Description: application/extension-pd
yeah, that's fun.
the logic behind the randomness is very good.
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So if you are 7.5m away from one of your band mates, there will 20ms of
latency just from the sound travelling thru the air.
if you are big enough to be playing on a stage that has 7.5m distance
between band mates, then you don't need to worry anyway, as someone else is
paid to sort out your
video works fine for me on ubuntu.
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i made a sound editor, if that's any use:
http://puredata.hurleur.com/sujet-1295-sound-editor
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honestly, one day i will figure out this 'internet' thing and upload all my
stuff to whatever svn or cvs or whatever it is that the pd-extended all goes
into.
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^
the mail above is also an example of why i don't want to release my stuff so
much.
all the patch is there, i am keeping no secrets. but still people expect me
to do all the work to add features to something which i gave them for free
and for which they can modify as they like.
i get the
i think you have to look at which method would benefit the greatest amount
of people. so, i think you'll find that far more people will benefit from
proper vanilla compatibility straight out of the box, than will benefit from
the cyclone library.
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vanilla pd should always have priority.
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What's more important, faster load times or less bugs? That's my
concern.
less bugs. every time.
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for filtering samples, try this: run an [env~] detector from the sample
player output, and smooth it with a [line]. then use that to modulate some
parameter of your filter, such as the resonance or frequency.
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or [getdir]
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Neko has been working on transferring the forum to a new server (or
something like that). I'm sure he is aware of the trouble and will get it
fixed as soon as he can.
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is it possible to do something vaguely similiar in vanilla pd?
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great, downloading now. i 'dropped out' of pd for about a year and got into
the oldschool hardware synthesizers, but recently i have picked up the ball
again with the digital stuff and am on the lookout for some inspiration like
this. cheers.
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embarrassed to say that i have been adding it myself all these years. i
even take care to make [PD] in capital letters.
it is information like this that needs more chapters in miller's books.
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just checking how 'clever' the server is
On Wed, Feb 3, 2010 at 8:29 PM, ypatios ypat...@gmail.com wrote:
embarrassed to say that i have been adding it myself all these years. i
even take care to make [PD] in capital letters.
it is information like this that needs more chapters in
i don't want a built in line~. there are plenty of synths i have where i
need to frequency to go to the correct value immediately. not to mention my
lfo's.
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when you think about it, in order to access data written, say 10 seconds
ago, then you need to have written that data in the first place. so, for
variable delaytimes, you have to choose the longest delay you will need, and
set your delaywrite to that length.
if you don't need realtime behaviour,
is there any need for the 1ms metro limit? couldn't it at least be lowered
to 0.01ms to enable sample accurate timing?
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yeah yeah, i know, so don't let it go to infinitely small values, but 1ms is
probably getting a bit outdated. surely it's time for a revision?
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i'm not looking for suggestions. i'm just saying that the 1ms lower limit
on metro speed is probably getting a bit outdated, and could be lowered to
0.01ms, to allow for sample accurate timing.
i assume that 1ms was an arbitrary decision made years ago when computers
were much less powerful, and
i agree with this suggestion too.
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that is a brilliant idea. if lines were always hidden behind boxes,
patching would be so much neater.
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are you any relation to roger moore?
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check out the help for [soundfiler] it can write as well as read.
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one thing i always use a lot in my pd patches is called a 'wetfilter'.
basically it is a bunch of bandpass filters, followed by delaylines of
increasing length. kind of oldschool 'spectral delay'.
so, i'd love to have this effect in a small hardware box, so i can put some
drum machines and
http://puredata.hurleur.com/sujet-1982-diy2-effects-sample-players-synths-sound-synthesis
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By the way I once collaborated with a guy who composes electroacustic music
and now is doing a Ph.D. at Harvard University (I made some Pd patches for a
piece he wrote)
i hope you got paid. because he certainly is.
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i also ran into this problem.
if i can remember correctly, the cause was something along the lines of:
the pd samplerate needs to be set before the delayline code is initiated.
if pd is run in gui mode, then the gui initialization forces the order
correctly.
however, when run in -nogui mode,
matteo, by putting a [switch~] in each patch, it could possibly alleviate
the problem, because i assume the [switch~] object forces samplerate to be
recalculated.
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So having a [switch~] would solve all the -nogui initialization
troubles? That is valuable to know. Thanks.
that was just a guess. we actually got around it by modifying pd code.
florian, sorry i can not post the link, because it wasn't me who did the
hack.
thanks roman, handy!
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