Thanks for the reply

My juggler just told me he didn't have no sound using the patch on Mac without no more info (I'll ask more because obviously it can be the dps which doesn't turn on or maybe just the soundfile location that doesn't have the same path on Mac, I can remember issues like this about path-shaping differing from Windows to Mac)

I'll give him some suggestions and shall be back with more info if I still can't find no solution

Thanks again

Hi there,

I've been patching this pretty simple metronome which has a few tempo preselection.
It's actually for a juggler I met, he had a very complicated setup to do something pretty simple during his show so I offered to take a few minutes patching this for him and send him.
My issue is that this works perfectly on windows (I only have windows and linux OS) yet his personal computer is a macbook and apparently it doesn't work there. I'll try as soon as I can get a hand on a macbook yet maybe some of yours might already have an idea about what's wrong ?

thanks guys

TheWhistle_GUI.pd

Hi there,

I begin to have quite an headache wondering how I can do that properly.

I mean
I get a signal from sensors, the sample frequency is 200Hz, and I want to make FFT and DWT analyse of this signal.

Yet the [FFT~] makes analysis from sound signal, so I convert my signal to a sound with [sig~]

It makes my signal from 200Hz to 44.1kHz, all right

The problem is that when I use [sig~] the signal doesn't put "0" between two samples, it keeps the value of the previous sample until it changes (so it's not like to raise the frequency rate and it modifies the spectrum...)

Any idea (the same problem with DWT)

because if I can get that I just have to put a low-pass filter after the upsampling and it's allright

Well the REAL question is "How can i upsample this signal properly from 200Hz to 44.1kHz "

Thanks

Let me explain the problem, I'm recording some values from accelerometers and from eyesweb.
Then I want to use it with scilab to make some treatment over it (FFT etc..) I know Pd could do FFT, but I need to do a lot of other stuff that Scilab does apparently better.

Yet I record the values that way :

5555;
6698;
658798;
etc...

my problem is the semicolon character, I did use it to terminate the message, when I've got it Scilab doesn't understand the variable the right way (and I can't find how to have Scilab to parse in real time fast enough)

So I'm looking for a way to record my file that way :

5555
6698
658798
etc...

I give the patch where I'm recording into the txt file "record_txt" (the right inlet just receive the name of the file that I parse from specific name and exercice during mesurement...)

Thank you very much,n
Yoha

http://www.pdpatchrepo.info/hurleur/record_text.pd

You should look at [FFT] object and here a good link with some more help

http://www.pd-tutorial.com/english/ch03s08.html

have fun

Well

I wanted to put zero in between to make the interpolation.
During upsampling the converters generally put zero between then there is a low-pass filter that makes the interpolation.

This is why I wanted to put zeros in between, yet you might be right, i'm gonna try with another software to make the interpolation the way you suggest and look if it does change the spectrum a lot, then if it works do it back in "real-time" with Pd

Then DWT is a "wavelet transform" it's quite close to a fourier transform but there are some very important differences
the base of projection is not made of sinus but of a "family of wavelet" that are some translated and dilated version of a "mother-wavelet"
It makes the transform able to work for signal which spectrum changes during time, and other kind of stuff (it is used for compression too, etc... very powerful and as long as IDS-analysis of Laurent Millot won't be able on Pd...)

This is a very useful way to analyse very low frequency (much better than fourier), and the phenomenon i'm working on seems to happen in very low frequency (between 0-20Hz )

I thought [block] just was up to decide the number of samples in a sound-block (then it makes you calculate FFT on different number of samples with the consequences on the precision of the frequency-axis) then I did look and ther is something about up and downsampling... gonna look too

Else you could try [moses], or maybe [clip] it depends on what you wanna do

Thanks a lot, I just didn't see this "cr" option, works very well
I was trying to do it with a "float2ascii" to make "\n" etc..