Hello all,

I'm having a hard time to achieve the following:

I have an audio signal that is 1 for 14 samples then 0, when it's "1" I'd like to output a "1" signal, then the next sample I'd like to output zero, in order to know at which sample the change occurred.

I want to avoid using "block~" to set audio block to 1 in a subpatch, and don't want to use externals.

Anyone has a clue about how to achieve this ?

Cheers

@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?

@jameslo At first I didn't use "lop~" thinking my signal was a constant "1".
I added the low pass and was able to detect attacks.

I'm now confronted to other problems, really hard to tell what's wrong...
I didn't post a patch at first because it uses a big external library ( flucoma ).

But after. a few days attempting to get the delay difference in samples between 4 mics receiving the sound of a snare drum I post here the ( buggy) patch I ended up with.
If anyone can have a look ( installing flucoma first ) I'd appreciate...

This patch is an adaptation for an artist who did this in Max msp , he needed this to work in puredate in order to use it with "Bela Gem" hardware.

I could post the "Max" patcher as well if needed.interOnsetDetection.zip![Screenshot 2026-05-07 at 11.38.55.png]

I attach also a screenshot of where the patch is at:

@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.

@jameslo Cheers,

I'm testing what I did to check if I get the onsets right.

the main difference with your exemple is instead of resetting the counter every other sample I need to reset it after the fourth onset is received .

another difference is the difference in between each onset is in the order of 4 to 20 samples.

Once I'm sure I get the onsets right I need to count from 0 for 1st Onset to 3 for the 4th onset and only then reset the counter and retrieve the values ( with samphold I guess )

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 ?

@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 Why not translate the (up and running?) Max patch 1:1 into Pd? (I can not see the Max screenshot.)

What is the overall latency-requirement?
And where do you output in the end, sigrate or controlrate?
Analyzing the blocks in control-rate is more common and easier but has at least 1 block additional latency, and can still be sample-accurate. (Sample-accurately one block late.)

reset the sample counter

When reset it? And why reset it at all?
Calculate difference by doing 'now minus before' without any reset.
But when has been before? i.e. what is Tzero? How do you decide which onset is now and which is before?

Do you just need to know which mic onset has been first or do you need inter-mic delay of all 4 mics? Things like that can be done by calculating the cross-correlation, but you have not asked for that, so I assume the Max patch is not doing that.

@lacuna

Why not translate the (up and running?) Max patch 1:1 into Pd? (I can not see the Max screenshot.)

Well this is what I'm doing, though my questions are about parts in max that are done in "Gen" ( hence my question about "detrending" a signal, the max patcher does it this in Gen ), there's a sample counter in the max version using "phasor~" , in Max the right inlet of phasor resets the phase with non-zero signal value, in order to reset the counter, in pd I can't figure out how to achieve this...

What is the overall latency-requirement?

the latency in the max patch is fixed, 256 samples.
Then as the distance from hits to each mic can't be more than 50 centimetres this latency could be dropped to 64 samples I guess ( @ 44100hz )

as for your last questions, "Tzero" is when the hit on the snare is received first, "before" is after the it's been received by all 4 mics.

Do you just need to know which mic onset has been first or do you need inter-mic delay of all 4 mics? Things like that can be done by calculating the cross-correlation, but you have not asked for that, so I assume the Max patch is not doing that.

the goal is to get the "cross-correlation" in order to calculate the position where the snare drum was hit, I think I explained this in a post above. Then my questions are about those max things that are not 1:1 transposable, and that got me lost a bit.

 (I can not see the Max screenshot.)

Would you mind helping me achieving this if I post a few screenshots of the max patcher ?
Let me know, I'd start a clean topic about this.

The initial question ( title of this topic ) has been resolved, but leading to other quirks I spent too much time on.

@lacuna Actually I could even attach the patcher, it would make it easier right ?

@ben.wes

EDIT2: and if it's about counting samples from the first hit to the others and then triangulating - that should also be achievable. let me know whether i understand correctly first. :)

Hi, I missed your post, but this is the idea. get the right number of samples from first hit to the others.
( The triangulation part is already sorted )

I should maybe start a fresh topic about this, as my original question was answered.