On Thu, 8 Apr 2010, Alexandre Porres wrote:
thanks, I have to correct the broken link now!cheers
do you mean http://artengine.ca/~catalogue-pd/28-Porres.pdf ?
Which broken link was that ?
Btw any links to http://pure-data.ca/ have to be changed to links to
http://pure-data.artengine.ca
thanks, I have to correct the broken link now!
cheers
2010/4/8 Mathieu Bouchard
>
> This was first presented in the Second Puredata International Convention,
>> in which I thank Mathieu again as well as the other people involved in the
>> production of it - the event was basically a turn point
This was first presented in the Second Puredata International
Convention, in which I thank Mathieu again as well as the other people
involved in the production of it - the event was basically a turn point
in my life for the better :) But the work I presented then seems to
have disappeared fr
Well, I ain't released at all, I just a briefly mentioned. But in a
portuguese mail list they were so excited about it I had to release and did
thank and make references to all of who helped, especially Mathieu, of
Course, without whom I'd still be stuck and not kowing what do to at all :)
And as
In Fact, I just compiled 2 ways of doing the phons curves, by
Robinson-Dadson, and the ISO226:2003.
glad to see it's released and GPL'ed, but can you just add « with lots of
help from Mathieu Bouchard » somewhere in the comments ?
_ _ __ ___ _ _ ___
Oops, couldn't find or load the [mtx_phon_curve]. But I got Pd Extended with
iemmatrix, what could I be doing wrong? do you all have it?
cheers
alex
2010/3/24 Alexandre Porres
> Hi, I found an acient thread that really interests me a lot!
>
> In Fact, I just compiled 2 ways of doing the phons
Hello all,
This abstraction is very nice, but computing matrices on every note is
quite expensive on the CPU, no? Perhaps it would be better to store it
in a table, let say using 60 (pd)db as a reference. The only thing is
that low and high notes are multiplied by factor much bigger than 1
and it m
2007/2/9, Ed Kelly <[EMAIL PROTECTED]>:
So, if you have a signal at 96dBPD (-4dBFS) and you set it to 90dBPD
(-10dbFS), the loudness will be heard to drop by the same amount as when you
then set it to 84dBPD (-16dBFS).
... but remember, that ear's frequency response curve is dependent on the
l
Hi Alexandre, hi all,
Yesterday I corrected an error in mtx_phon_curve-help.pd and changed the
behaviour of [mtx_phon_curve] abstraction
(pure-data.cvs.sf.net/cvsroot/puredata/externals/iem/iemmatrix/abs).
Now the output is in dB, just like everyone would expect; before it was
p^2/p0^2 ...
On
Hi Alex, all
...but PDs dB goes to 0, because we have to have the numbers following the same
scheme - so that 0 means 0 always I think. Industry uses dBFS (deciBel Full
Scale) because it is measuring the signal against the full scale from -96dB in
16bit recordings to 0dB (all bits used).
Ba
Hi all,
2007/2/8, Frank Barknecht <[EMAIL PROTECTED]>:
Yes, it's new. You could use the [list-len] abstraction from
[list]-abs instead, but you need to check out an older version,
because the newest one just wraps [list length].
zexy's [length] also does the job, as far as I know.
As I can se
Hallo,
Alexandre Quessy hat gesagt: // Alexandre Quessy wrote:
> It appears that the object [list length] cannot be created. Is this a
> pd 0.40 new feature of the [list] object ? (using pd 0.39-2 extended
> on Linux)
Yes, it's new. You could use the [list-len] abstraction from
[list]-abs instead
Hello all,
Very nice, mr. Zmoelnig. Of course, this is exactly what I need.
It appears that the object [list length] cannot be created. Is this a
pd 0.40 new feature of the [list] object ? (using pd 0.39-2 extended
on Linux)
To use this very nice abstraction, I simply need to multiply the
amplit
Alexandre Quessy wrote:
> Hi all,
>
> I would like to create an abstraction to adjust the amplitude of
> synthetic sounds according to the ear sensitivity. At a given
> amplitude, we hear the notes in the middle range louder than the high
> and low notes. This perceptual property of the audition c
Hi, Alexandre,
On the one hand, a filter could be designed to approximate what you
need, but this sounds like a job for the wavelet transform. A linear
filter can't handle the level-dependent changes in iso-sonic curves.
In the wavelet domain, you could apply different compression/expansion
curv
Hi all,
I would like to create an abstraction to adjust the amplitude of
synthetic sounds according to the ear sensitivity. At a given
amplitude, we hear the notes in the middle range louder than the high
and low notes. This perceptual property of the audition can be
somewhat undesired in a music
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