• seb-harmonik.ar

    @Joseph-Mikkelson sorry, someone that knows about windows will have to help.. but generally you have to edit PD_PATH to point to the correct location of the pure data executable..

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  • seb-harmonik.ar

    phasor~ is "dirty" because is not anti-aliased (or band limited), so harmonics will "fold down" back onto the spectrum - there are many different methods of antialiasing oscillators.. there are none in pd-vanilla though.. I have some in my library using stilson & smith's "blit" (which doesn't work so well for frequency modulation though) https://ccrma.stanford.edu/~stilti/papers/blit.pdf

    the miniwoog uses mmb's band limited saw https://github.com/dotmmb/mmb, which uses pre-computed band limited waveforms and crossfades through a large table of them and selects the appropriate number of harmonics based on the fundamental frequency.. this is a very safe method but takes more memory afaik

    there are polyblep-based band limited oscillators in heavy lib https://github.com/enzienaudio/heavylib
    and also mini blep ones in creb

    posted in technical issues read more
  • seb-harmonik.ar

    @Joseph-Mikkelson no, the makefile should work for Windows if your system is set-up for building pd externals (I'm not quite sure what that entails..)

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  • seb-harmonik.ar

    l2ork has an autotune add-on that works ok imo (& can be compiled for mac or windows as well)
    https://github.com/pd-l2ork/pd/tree/master/l2ork_addons/autotune

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  • seb-harmonik.ar

    @jenkutler that's where I built it from

    posted in extra~ read more
  • seb-harmonik.ar

    @dadasane yeah the easiest way is to sample every x samples without any filtering.. Of course, you can only reliably get "predictable" aliasing at integer divisions of the sampling frequency afaik.. you could also upsample first, (and use an upsampling filter) and then decimate (downsample) in order to get more resolution

    if you want to sample at other frequencies besides integer divisions of the samplerate, you have to do some kind of resampling - figure out the "correct" inter-sample value, then interpolate back to the original sample rate by low passing the hypothetical "sample-and-held" signal below the nyquist frequency

    posted in technical issues read more
  • seb-harmonik.ar

    @boonier here's the "single binary" version:
    percolate.zip

    (you have to load it as [declare -lib percolate], or put it in the startup flags)

    posted in extra~ read more
  • seb-harmonik.ar

    "sleep"ing refers to system calls on many (all) systems to yield control of the process to the operating for a certain amount of time.. basically it freezes (delays) the process.. however "spinning" is where the process still technically does stuff and yet does nothing (like a "while" loop) just to not yield control over to the operating system's scheduler afaik.. so that it doesn't do anything unexpected or unnecessarily delay the process returning to activity.. I think. (Of course this uses resources/computing power/energy that would otherwise be used on other stuff)

    But "sleeping" is generally used if the scheduler has done all of the work it's supposed to do and is waiting for the dsp tick to finish/ time to pass until the next 64 samples I believe..

    As for the rest of the questions.. I'm not sure but I think that messages are computed every 64 samples no matter what. Objects like vsnapshot~ and vline~ do what they do through getting the logical time of messages they receive (which happens in-between when the audio is computed), storing that, and then using that logical time converted to samples in their perform routines.

    As for why it's more expensive to call things one sample at a time: Mainly bc there is an overhead associated with function calls and I think. Every time you call a function system resources and space on the stack are used. So if you minimize function calls by processing signals in batches it's computationally cheaper

    posted in technical issues read more
  • seb-harmonik.ar

    @forrestbaer I've had this problem with tcl/tk 5 compiled for mojave.. did you compile it yourself or are you using miller's tcl/tk? Using the supplied tcl/tk or one from before mojave should work ok in my experience..
    see this thread from the mailing list: https://www.mail-archive.com/pd-dev@lists.iem.at/msg02102.html

    posted in technical issues read more
  • seb-harmonik.ar

    @ddw_music here https://github.com/pure-data/pure-data/tree/master/extra/bob~ miller refers to the "cutoff frequency" which would be an odd term to use for a bandpass filter.. if you look at bob~ without any resonance it definitely behaves like a a lowpass.. I think that as the resonance goes up the pass-band might get attenuated? most discussions of the moog ladder filter refer to the lowpass filter (rather than the High-pass one) and since this certainly is not a high-pass filter I would say that in all likelihood it's an emulation of the lowpass species.
    If you want to go for instructional clarity I agree that maybe other filters look more like "lowpass" filters in spectrograms tho (but bob~ still sounds like a lowpass to me imho)

    posted in technical issues read more

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