@jorbecke I am not qualified to opine on your mental health, but if you are referring to the hurleur links going nowhere, then it might be a Google Chrome thing. Is that what you're using? If so, try another browser. Otherwise I'd have more modest expectations of a 13 year old thread

@y0g1 My friend who uses Max/MSP just explained to me that phasor~ resets when the right input signal changes to a non-zero value, which means that in my fexpr~ version the test should be "$x != 0 && $x != $x[-1]"

Edit: argh, I even misunderstood my friend. @lacuna is correct below--it's when the signal changes to a non-zero value from zero

@y0g1 said:

"phasor~"'s phase is reset when receiving a non zero signal.

If this was a test in college I'd be there during office hours arguing that the question was misleading

@y0g1 Assuming the sample frequency is 44.1khz, the phasor frequency is fixed at 1, and you don't need to support any other feature than what you mentioned above, try [fexpr~ if($x != 0, 0, ($y + 1) % 44100)] to replace both the phasor and following multiplication.

@playinmyblues The object named a12 should be an array, not a canvas. Replace it and things should start working.

@y0g1 said:

How would you adapt metronome timer for getting the difference between 4 consecutive "onsets", then reset the sample counter to be ready for the next 4 onsets ?

Ugh, in audio domain? I dunno but off the top of my head I think it would be nasty/not worth it. Can't you just take a stream of timing intervals in control domain and process them in groups of 4? How you would align each group of 4 isn't obvious to me either, which is what I think @ben.wes was alluding to earlier RE "cooldown period". Edit: maybe [fexpr~] could help.

@y0g1 I'm unable to test, but your patch looks like [fluid.ampfeature~] and [fexpr~] is supposed to be outputting 1 when the signal is above some threshold and 0 otherwise, is that correct? Is it working as you expect?

If so, the only thing you need from my patch is the part that extracts the leading edge as a pulse, inverts it, and then counts the samples between pulses, but if the signal out of [fexpr~] isn't clean (i.e. outputs one and only one contiguous block of 1s for every drum note) then my counter will output junk. That's why I was asking about [lop~] and threshold in my patch, which is what I assumed you were using for signal conditioning, e.g. for novelty metronome sounds I had to adjust them to get clean square pulses.

@y0g1 What value are you using for the threshold? What frequency are you using for [lop~]? Can you please post a recording of several snare hits?

@donnerbono Yes, and you can even change the array at the same time, but one action, either the read or the change, will come first and will therefore make things sound differently. It's not obvious to see which will come first, which is a dead horse I've beaten previously

@y0g1 The formula for [rzero~] in help means "the output is the difference between the current input and (the previous input times the current coefficient)." Since the coefficient is always 1, it's just the difference between the last two inputs. So since the input is just a square wave toggling between 0 and 1, the output is going to be 1 on the rising edge, -1 on the falling edge, and 0 otherwise. By clipping it to 0..1, I just take the rising edge pulse.

"Convert to negative pulse" means instead of sitting at 0 with the occasional 1 pulse, it now sits at 1 with the occasional 0 pulse.

The formula for [rpole~ 1] in help means ""the output is the sum of the current input plus (the previous output times the current coefficient)." When both the input and coefficient are 0, the output is 0, and when they're both 1, it increments by 1 each sample (i.e. counts samples).

By now you probably know how to understand the formula for [rzero_rev~ 0], let me know if you don't. The reason you need it is so that [samphold~] grabs the highest previous count, not the reset value.