Atte André Jensen wrote:
patrick wrote:
1) pd -? says -audiobuf is in ms, so 256 wouldn't make sense, or...?
why shouldn't it make sense? 256ms is justas valid as 7ms
2) I thought that jack clients automatically got the same buffer size as
jack, which means that is doesn't make sense
Matt Barber wrote:
Any ideas??
have a look at how iemlib's filters are implemented. most of them are
really just abstractions wrapping the correct parameters for a generic
implementation class for the non-maths among us.
fgamsdr
IOhannes
___
IOhannes m zmoelnig wrote:
Atte André Jensen wrote:
patrick wrote:
1) pd -? says -audiobuf is in ms, so 256 wouldn't make sense, or...?
why shouldn't it make sense? 256ms is justas valid as 7ms
Because he's building a commandline to match a jacksetup of
Frames/Period: 256...
specify that
Atte André Jensen wrote:
IOhannes m zmoelnig wrote:
Atte André Jensen wrote:
patrick wrote:
1) pd -? says -audiobuf is in ms, so 256 wouldn't make sense, or...?
why shouldn't it make sense? 256ms is justas valid as 7ms
Because he's building a commandline to match a jacksetup of
Hi
What's the best way to do 1/x? I have something like:
|10 (
| \
|bang( |
| /
|1(/
|/|
--
peace, love harmony
Atte
http://atte.dk | http://myspace.com/attejensen
http://anagrammer.dk | http://modlys.dk
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[expr 1/$f1]
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a good place to find different kind of audio interpolator :
http://www.student.oulu.fi/~oniemita/dsp/deip.pdf
cyrille
Matt Barber a écrit :
For polynomial interpolation using four points, if the above is right
there are 5 ways to do it, and they are ordered first by degree of
polynomial,
Hallo,
Atte André Jensen hat gesagt: // Atte André Jensen wrote:
Hi
What's the best way to do 1/x? I have something like:
|10 (
| \
|bang( |
| /
|1(/
|/|
Some more are here:
http://puredata.info/docs/tutorials/TipsAndTricks#reciproce-a-number
Ciao
--
Frank
On Fri, 27 Jun 2008, Charles Henry wrote:
I get what you're saying now. I had to read it a couple times through
to see :) You're referring to piecewise cubic polynomials, right?
Yes, I'm always assuming that piecewise-cubics is all that we'll need.
We would wind up with an overdetermined
[pow -1] seems to be simple.
++
Jack
Le 2 juil. 08 à 09:43, Atte André Jensen a écrit :
Hi
What's the best way to do 1/x? I have something like:
|10 (
| \
|bang( |
| /
|1(/
|/|
--
peace, love harmony
Atte
http://atte.dk |
On Sat, 28 Jun 2008, cyrille henry wrote:
i personally consider that the interpolation should not add harmonics,
and should remove non audible harmonics. i.e : a noise with freq from
20Hz to 20kHz shift 2 octave lower should result in a noise with freq
from 20Hz to 5KHz. but it's ok for me if
On Sat, 28 Jun 2008, Matt Barber wrote:
a1 = 0.5f * (c - a);
a3 = 0.5f * (d - a) + 1.5f * (b - c);
a2 = a - b + a1 - a3;
*out++ = ((a3 * frac + a2) * frac + a1) * frac + b;
10 +'s 6 *'s
If you compute twice the value of a1,a2,a3 and later multiply by 0.5, you
end up with a multiplication
Hi,
On MAC OSX 10.3.9 with GEM 0.90 video and sound play sync. (PD 0.38.4
extended-RC7)
Carlos
On 08/07/01 16:05, Andre Cardozo [EMAIL PROTECTED] wrote:
Hello all,
I´m working on a project which requires some videos to be displayed on
screens, along with their corresponding soundtracks.
Andre Cardozo wrote:
Hello all,
I´m working on a project which requires some videos to be displayed on
screens, along with their corresponding soundtracks.
I do know how to use GEM to display and manipulate the videos, but as
far as I know GEM ignores the embedded audio tracks, as is a
= RJ Sprint, July 2008 =
www.realityjockey.com
= Kickstart interactive music =
RJ is an interactive music format for the iphone and other mobile
devices. The goal of the project is to turn interactive music into a
consumer format.
? RJ tracks are mainly consumed with headphones. Think of
recently i have discovered the joys of using overlapped blocks in sound
patches. however, i'm having a little bit of difficulty understanding
exactly WHAT gets overlapped, and what doesn't get overlapped.
it seems to me that control messages and do not get overlapped, and i can't
really tell
PSPunch wrote:
[phasor~] takes a range of 0 to 1.
To fully drive your speakers, you may want to insert a [-~ 0.5] after
[phasor~] so that you get a range of -0.5 to 0.5, centered at zero.
You can then multiply the amplitude by 2 (instead of 1 which you
currently have your slider range set
if you use overlapped blocks, it makes the hanning window less intrusive.
so instead of [block~ 512], use [block~ 512 2 1] or even [block~ 512 4 1]
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uhm...
actually, *not* using the hanning window, makes the processed sound suck
even more.
so what's the best solution?
there is a better window function? perhaps the gaussian function?
isn't the window function also related on how much samples overlap
between a block and the next one?
# If the block size is N samples (and there is no overlapping) then the
patches (subpatches) calculate the audio stream after every Nth sample. At
every tick the inlet~s take N samples in and the outlet~s give N samples
out.
# Overlapping is like you have M identical audio streams which are N/M
On Tue, 1 Jul 2008, Matt Barber wrote:
Any ideas??
Just drop the idea of matching more than two sample points. It's what
makes [tabread4~] miss the opportunity to be C1, but it's also in exchange
for pretty much nothing. Well, maybe it's not nothing, but I still have no
clue about what's
Hallo,
[EMAIL PROTECTED] hat gesagt: // [EMAIL PROTECTED] wrote:
today I start learning FFT, and after seeing the (hann) windowing
function, I realized this (attached) filter with custom frequency
response, but I suspect something is wrong here
why it sucks (given that it does)?
First as
Hallo,
PSPunch hat gesagt: // PSPunch wrote:
The easiest way to avoid this that I know of is to clone your entire
FFT routine (call them 'original' and 'clone' for now), apply a delay
before the original signal and the same delay time after iFFT on cloned
signal. The delay time should be
Hallo,
Atte André Jensen hat gesagt: // Atte André Jensen wrote:
What's the best way to do 1/x? I have something like:
|10 (
| \
|bang( |
| /
|1(/
|/|
As far as I see it, here you may have an execution order problem, at
least the first tim you bang that ascii
Hi list.
Is there any way to make an array in a data structure *not* responding to
clicks ?
Cheers.
_y
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On Wed, Jul 2, 2008 at 1:26 PM, Mathieu Bouchard [EMAIL PROTECTED] wrote:
On Tue, 1 Jul 2008, Matt Barber wrote:
Any ideas??
Just drop the idea of matching more than two sample points. It's what makes
[tabread4~] miss the opportunity to be C1, but it's also in exchange for
pretty much
Frank Barknecht wrote:
As far as I see it, here you may have an execution order problem, at
least the first tim you bang that ascii art. ;)
Is that because of depth first, which would mean that / sees the 1 at
hot inlet first and then carries out 1/0 before the 10 arrives at cold
inlet?
--
It shoud be that way
in
|
[t b f]
| |
[1( |
| |
[/]
|
out
but other possibilities are better in my opinion.
Anyway, in the link that Frank gave you is better explained.
_Ricardo
2008/7/2 Atte André Jensen [EMAIL PROTECTED]:
Frank Barknecht wrote:
As far as
Hallo,
Atte André Jensen hat gesagt: // Atte André Jensen wrote:
Frank Barknecht wrote:
As far as I see it, here you may have an execution order problem, at
least the first tim you bang that ascii art. ;)
Is that because of depth first, which would mean that / sees the 1 at
hot inlet
Frank Barknecht wrote:
Yes: Everytime you have connections fanning out of a message outlet,
the order the connections fire is not specified. It's practically
undefined.
Ok, that seems like something that's possible to identify (with practice).
I'm a little confused about loadbang, then.
Hi
Is it possible to reset phasor, so sending it a message that'll make it
start from 0 again?
I tried using a metro/vline substitute, but couldn't figure out how to
change the speed *within* a cycle. Is such a construction possible?
NB: I'm trying to read out a sample from a table...
--
Atte André Jensen wrote:
I'm a little confused about loadbang, then.
Or maybe not. I guess it's exactly the same...
--
peace, love harmony
Atte
http://atte.dk | http://myspace.com/attejensen
http://anagrammer.dk | http://modlys.dk
___
Hi,
as i remeber from the phasor's helpfile, sending a bang to one of its
inlets resets the phase to zero.
rgrds,
p8r
* Atte André Jensen [EMAIL PROTECTED] [2008-07-02 23:45]:
Hi
Is it possible to reset phasor, so sending it a message that'll make it
start from 0 again?
I tried using a
Hi
You can set the phase through the right inlet. Just send a zero and the
phasor starts a new cycle.
alabala
2008/7/2 Atte André Jensen [EMAIL PROTECTED]:
Hi
Is it possible to reset phasor, so sending it a message that'll make it
start from 0 again?
I tried using a metro/vline
Peter Plessas wrote:
as i remeber from the phasor's helpfile, sending a bang to one of its
inlets resets the phase to zero.
Sending a float to right inlet resets phase (it seems regardless of the
value of the float).
I'm not only newbie, I'm also blind :-) Sorry, about that!
--
peace, love
Peter Plessas wrote:
as i remeber from the phasor's helpfile, sending a bang to one of its
inlets resets the phase to zero.
Sending a 0=float=1 to the right hand inlet sets the phase at the next
block boundary (usually 64 samples).
* Atte André Jensen [EMAIL PROTECTED] [2008-07-02 23:45]:
I
changing speed within a cycle using vline:
[metro 1]
|
| [r to-float]
| |
[f ] [enter your new speed here(
| |
[t f f][* 44.1]
| | |
| [+ 44.1]
| |\
| | [s to-float]
| |
[pack]
|
[$1, $2 1(
|
[vline~]
|
[tabread4~]
try setting the environmental variable GEM_SINGLE_CONTEXT to 1
before starting pd/Gem and tell us what happens.
still crashes and reports:
GEM: stop rendering
socket receive error: connection reset by peer (104)
Segmentation fault
___
i'm trying to make the eeePC (xandros, 1gb ram) to be the genius setup that
i envisioned. first, i am aware this is a limited machine, but i plan to do
basic synthesis on it. nothing too complex.
i'm hiding as much GUI elements as i can in subpatches, i start pd with -rt
flag, and i turned the
On Mon, Jun 30, 2008 at 06:10:46PM +0200, Atte André Jensen wrote:
Claude Heiland-Allen wrote:
in~ [phasor~ 4410]
| |
[samphold~]
|
out~
Ah, of course!
Note: this doesn't change the sample rate, but resamples the signal at
4410Hz. Not sure if this is what you want, but
Hello,
There are applications I want to use on Linux that speak only jack
midi and not alsa seq. Many of the softsynths I like, however, still
use the alsa seq interface. Is there a way to make pd act as a glue
between these two? I am not a programmer, and to be honest,
complicated things
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