monohusche

what's the most efficient way to get from a signal in the msp domain to discrete steps ?

signal (0..1) --> [snapshot~ 1] --> [* 16] --> [change], but I would think taking such frequent snapshots isn't efficient.

efficient-way-to-generate-step-sequence-from-phasor

Quote: monohusche wrote on Wed, 28 January 2009 15:38

----------------------------------------------------

> what's the most efficient way to get from a signal in the msp domain to discrete steps ?

> signal (0..1) --> [snapshot~ 1] --> [* 16] --> [change], but I would think taking such frequent snapshots isn't efficient.

I'd be happier with a bunch of >~'s and edge~, too, as per X37V.

I wrote a thing called Stepmetro that's a modular implementations of this approach. It's in the CNMAT MMJ Depot:

Quote: monohusche wrote on Wed, 28 January 2009 15:38
----------------------------------------------------
> Hi all,
>
> what's the most efficient way to get from a signal in the msp domain to discrete steps ?
>
> My approach: 
>
> signal (0..1) --> [snapshot~ 1] --> [* 16] --> [change], but I would think taking such frequent snapshots isn't efficient.
>
> thx
----------------------------------------------------

Not sure what you have planned for the triggers, but if you wish to stay entirely in the signal world, you might enjoy the MSP externals I posted on my share page:

https://cycling74.com/twiki/bin/view/Share/AndrewBenson

Not sure what you have planned for the triggers, but if you wish to stay entirely in the signal world, you might enjoy the MSP externals I posted on my share page:
https://cycling74.com/twiki/bin/view/Share/AndrewBenson

If you need audio-accurate sampling times, you may sample the signal on an audio clock with sah~, using the accumulated clock pulse as an index to poke~ a buffer. Then dump the buffer as Max events with uzi/peek~. Using a stereo buffer and a flip-flop logic, you can dump channel 1 while channel 2 is written.

If event sampling rate equals audio sampling rate/2**n, you may also use poke~/peek~ inside a downsampled poly~ and dump every sample.

this sound very interesting. I haven't seen this way before. Can you post an example, please?

hi j.y.bernie,
this sound very interesting. I haven't seen this way before. Can you post an example, please?


you couls also use a seq~ with the folLowing cue in it:

On Jan 28, 2009, at 18:38, Nick Laqua wrote:

> what's the most efficient way to get from a signal in the msp domain 

> signal (0..1) --> [snapshot~ 1] --> [* 16] --> [change], but I would 

> think taking such frequent snapshots isn't efficient.

0, id sixteenths;
1, 0. 1;
2, 0.0625 2;
3, 0.125 3;
4, 0.1875 4;
5, 0.25 5;
6, 0.3125 6;
7, 0.375 7;
8, 0.4375 8;
9, 0.5 9;
10, 0.5625 10;
11, 0.625 11;
12, 0.6875 12;
13, 0.75 13;
14, 0.8125 14;
15, 0.875 15;
16, 0.9375 16;

>
> Hi all,
>
> what's the most efficient way to get from a signal in the msp domain  
> to discrete steps ?
>
> My approach:
>
> signal (0..1) --> [snapshot~ 1] --> [* 16] --> [change], but I would  
> think taking such frequent snapshots isn't efficient.
>
> thx


This solution requires cambio~ from cnmat (

This solution requires cambio~ from cnmat (http://www.cnmat.berkeley.edu/MAX).

Quote: i.te wrote on Thu, 29 January 2009 09:21

> I haven't seen this way before. Can you post an example, please?

I know no simple way to pass from the signal domain to the events domain with enough time-accuracy, since both schedulers are distinct. Sampling with a Max clock (snapshot~) will not produce evenly spaced audio samples, unless Max Scheduler in Overdrive is on AND Scheduler in Audio Interrupt is on AND Signal Vector Size is low enough.

You may try this: sample a LFO with snapshot~ 1. Use the output to FM modulate a cycle in the audible range. The tone "copies" the LFO only when above DSP conditions are met.

cambio~ does not depend on Overdrive/Interrupt. From what i can see (and hear), it outputs 3 evenly spaced audio samples per signal vector. So, you are required to set Signal Vector Size = 3*(Sampling Frequency)/(events/sec you want).

It all depends on your application, how much events/sec you want, and how much time accuracy relatively to the audio clock you need.

There are two steps involved: downsampling and converting audio samples to events. As I needed both high timing accuracy and precise event rates down to millisecond to convert digital modulations to bits, I've tried using poke~/peek~ for the second step. Downsampling may be accomplished with sah~ (arbitrary sampling rate) or with poly~ (audio sampling rate / 2**n).

None is really satisfying since we need, at some point, to call Max's scheduler (edge~). Again, there is a dependency on vector size, Overdrive and/or Interrupt. However, that was enough for me to get proper bit stream decoding (ever tried to program an UART in audio?). Here is the poly~ example. Wrap in a poly~ and send the message 'down N':

Quote: i.te wrote on Thu, 29 January 2009 09:21
----------------------------------------------------
> I haven't seen this way before. Can you post an example, please?
----------------------------------------------------

efficient way to generate step sequence from phasor