can't get a phasor to bang as accurate as i want
as sugested i created a new topic for my problem with hopefully a more clear topic title.
is there a way to make a patch like this more accurate?
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normally i use a signal vectorsize of 64 (and an I/O size of 256 but i don’t think that matters?). i thought that was pretty small but even with a phasor of 1 hz it causes inaccuaracy of @ highest 2 ms. does it time that way with that vector size on your computers too?
when i use a signal vector size of 16 an a phasor of 1Hz the intervalls are rock solid but when i calcute in example 145bpm wich makes phasor go at 2.41667Hz the intervalls switch beteen 413.666656 an 414 ms. even with a vector size of 2 the timed intervalls sporadicly switch from 413.791656 to 413.83334 wich is a vector size i think i couldn’t affort.
is this normal or is it only my computer?
i tried using the internal souncard but this gives me even more inaccuracy on all these settings i tried with my other soundcard, wich as a ‘focusrite saffire pro10’, i aslo tried my ‘m-audio fasttrack pro’ wich gives the same results as the focusrite.
i also can’t find any background processes wich use ramrkable much cpu or memory.
strange thing is that when i use a metro at a vector size of 64 at whaterver floating point interval(tried 4n with 145bpm, 414.75689 ms, and just 100 ms) it is rock rock solid.
In what context is this behavior problematic? It could be the case that you don’t have to leave the signal domain at all. Without context this is hard to judge.
i’m using a quantization technique with onebang to start and stop record~, start and stop play~ and time the lenght of the recorded sample to drive the phasor wich drives the play~ object.
If the start and stop commands are provided by a user, then there is no need to worry about (fractions) of milliseconds, because users are typically less accurate then that. Are the recording and playing separated in time? Still it is unclear where according to you timing inaccuracy becomes intolerable. Until you provide some sort of an example (even a sketch, you don’t have to give it all away) it is impossible to come up with a remedy.
I’m actually trying to do a similar thing using poke~ and index~ for a looping patch.
I’m using the length between the start and end of recording to determine the speed of one master phasor~ for normal speed playback, and trying to sync all overdubbing to the phasor~, with the overdub "quantized" so it starts automatically in sync with the playback, and also stops automatically. I havn’t yet worked out how to get it to stop by itself, and I’m running into sync issues between each overdub (maybe aliasing as well – not too sure but it sounds terrible).
Heres the link to the thread, there are a few patches I’ve posted and lots of suggestions from others on how to sync things using phasor~, so you might find something helpful.
I’m trying to stay away from edge~ etc… to keep my whole patch signal controlled, so that the sync stays accurate, I’m not sure how accurate you will be able to get if you base the patch around record~. But, then again, I’ve not managed to get mine working either!
thanks for helping. i would only like to know for now if the things i write about in the first post of this topic are only appearing on my computer or on yours too?
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