## Difference in float and signal

Jun 08 2010 | 5:13 pm
Hello Max/MSP forum!
I wonder whether someone could teach me here something? What and why is there a difference between these two signals if both have the same source (code below).
Explanation appreciated! All the best Krisztian
code:

• Jun 08 2010 | 5:27 pm
well one is coming straight from a cosine wave oscilator, the other is just a the same osc resampled (at 50Hz, one sample each 20 ms), therefore it's not continuous anymore.
• Jun 08 2010 | 5:41 pm
I replaced your spectroscope~ objects with scope~ objects so you can kind of see the sample-and-hold situation EJ describes.
• Jun 08 2010 | 5:59 pm
this is interesting... the [number~] object only does 20ms interval update. Is there a way to resample into a continuous signal and if not why?
thank you for both! K
• Jun 08 2010 | 6:21 pm
If you want to resample, that probably means that you want to loose samples. Converting from the signal domain to the control domain will result in lost of precision, because you can't get 44100 values per seconds (assuming 44.1kHz is your sample rate), that would be really inefficient. It's hard to answer really because it's unclear what you want to achieve though ;-)
• Jun 08 2010 | 6:49 pm
sorry, my English teachers were not strict with me:)
Your reply was needed to give me an idea how to proceed in testing two different EEG analysis approaches, one that uses the 'classic-vocoder example' and another one with the [ffb~] object. I think I have to sit down and study DSP from the basic. thanks again, and all the best, Krisztian
• Jun 08 2010 | 8:30 pm
hello again, It is quite crucial to understand the following. Please 'pass or fail me' on this DSP issue:
I receive 120 samples in a second from the serial port (EEG - 120hz). Do I HAVE to make this 'float' (-0.0019859, -0.02141, ....) into a 'signal' in order to use (1) digital filters or (2) FFT?
If the answer is yes, what exactly does sig~ do? When it increases the sample rate (120hz to 44.1kHz) does it add 43.920 samples by connecting the original ones (120)?
Please let me know whether I am right or wrong. Apologies if this is really silly :)
all the best K
• Jun 09 2010 | 7:35 pm
Hello the Dude
It is only analysis of EEG, where I'd like to use the energy of certain bandwiths. With 120hz sampling rate the changes in the signal are not too gradual, but I guess for analysis it should be OK. As there is no intention to sonify, to make the signal audible, I assume we do not need interpolation.
"first cycle at 120Hz could input 42.3 and so the 400 samps(at 48k) will all be 42.3, then the next cycle..."
A float, I can imagine as being an array of numbers. The signal must be a bit more complex, although [number~] outputs a float like value. In the above line, what does it actually mean, when you say "first cycle inputs 42.3..." ?
I reckon it creates a sine wave of 42.3hz and this freq stays for an other 400 samples (at 48k). Is this close to the truth?
Thank you, this helps a lot. all the best Krisztian
• Jun 09 2010 | 9:21 pm
aha! I think I am getting it:
If I use [cycle~ 440], it outputs a float from -1.0 to 1.0, 440 times in a second ! I assume, as the software is digital, the sine wave or ANY SIGNAL is digital too, and therefore it is "an array of samples storing floats". The main difference between the float and a signal is that signal has a higher sample rate.