LFO vs Mini LFO

Something really weird (and cool) happens when you use LFO as a unipolar saw to drive the Offset of Flexi in Scratch mode. There is some aliasing seemingly in the LfO signal that gives you weird pitch shifting artifacts from Flexi. Sounds not unlike a vintage sampler. You do not get this aliasing when you use Mini LFO or a Graphic modulator.


  • I'm gonna check this, thank you

  • @giku now an vintage pitch shift algorithm on sampler would be nice :) !

  • +1 with option to disable the linear interpolation, can’t help it I love the rough edges when transposing down short samples a lot (2-4 octaves or more) by repeating (sample & hold) samples instead of smearing them with linear interpolation which makes them sound muffled and mushy instead of sparkly and airy…

  • I use bicubic interpolation.

    yep there may be such an option but that's what Decimator is for

  • Yelp, but to have accurate keyboard scaling for the decimator can get a bit tricky.

    I've tried my best to set the decimator frequency to 130Hz to match the oscillator tunings and then use the note value and scale to the max so the sample-rate scales proportionally with the played note. This gets me roughly in to the wanted ball-park.

    The interpolation options in SunVox and Renoise are nice and the adaptive oversampling option Renoise provides is a nice bonus and going strictly by ear the adaptive oversampling allows a very wide transposition range upwards without major artifacts ruining the sound apart from the sound getting shorter and it works very well with looped sounds.

    I know Drambo is by design a 'modular groovebox' but in some cases it becomes a chore...

    (I'm really looking forward to the new 5OP FM2 module in SunVox 2.0 as it's quite tricky to implement an 'algorithm change' in Drambo).

    And even the current 2OP FM module in SunVox is 'instant gratification'.

    I love Drambo for what it is already and as with everything once the candy is there we all want more 😎


  • edited January 2022


    I use the decimator as some kind of lowpass filter that does do aliasing

    its really cool (and musical) with key follow 😃

    what are other interesting ways to use the decimator?

  • No limits! Be playful and find out for yourself 😉

  • Just want to add that I too love different interpolation modes for samplers! Never really been able to reproduce the same kind of crunchyness with decimators.

  • @rs2000 thats the spirit!

  • I will agree with those asking specifically for sample interpolation options though…I think the magic of many of the old samplers came from more than just decimation of the signal. It’s not a priority for me but a SP1200 algorithm for pitching samples in Flexi would be amazing.

  • edited January 2022

    12 bit plus decimator plus analog filter is in that ballpark (aliasing was unwanted so they filtered it down)

    what is missing is that crude Pitch algo (that I don’t miss)

    btw I think it was just one octave up down?

    So not that important as u don’t get that let’s pitch it down 4 octaves thing that other stuff did (mirage or what have you …)

  • I can’t really hear much effect from the bit redux until I get to under 6 or so bits. Not sure if my ear is just off or what.

  • edited January 2022

    hehe your ears are fine

    slave to the rhythm was a 12 bit master

    you would never guess if you don’t know that

    if you don’t have really quiet sounds like a jazz trio or something you don’t notice the 12 bit at all ;)

  • Regarding sample interpolation...

    Here's an example in SunVox (It has 3 options for interpolation, Off, Linear and Spline).

    In this example I'm just using a simple HiHat sample...

    First run us 'Spline', after that 'Off','Linear' and 'Spline'

    It's quite easy to hear how 'airy' the 'Off' version without interpolation is and it gets more 'muddy' with Linear and Spline.

    It get's equally interesting when transposing up with with no interpolation as various aliasing effects ranging from sandpaper to more solid sounds.

    I'd say that the actual interpolation method plays equally big roll in the 'character' of the sound as reducing the number of bits and bsamplerate does.


  • edited January 2022

    Hm, what I hear is its falling apart with varying levels of nastyness when pitched down

    im not not sure what u mean with air - more aliasing?

    btw. If you just play a single pitch that’s all ok

    but if you want to play a melody with that you play notes up the keyboard and the aliasing comes down to you at the same time, that’s very unmusical 🧟‍♂️

  • Air is the 'vertical & horizontal gap' between samples when they are not 'interpolated' ie. first sample value is 1 and the second sample is 10, so instead of interpolating the values (2,3,4,5,6,7,8,9) the output value stays the same until it's time to fetch the second sample.

    This is partly how the old samplers did pitch-shifting (Amiga's Paula DMA is one example) they set the output value to the sample value and held it for a duration until it was time to fetch the next sample, the 'duration' defined the playback pitch, the shorter the duration the faster the sample played back and when things needed to be pitched up above the maximum playback sample rate samples were simply dropped (so 1 octave up meant every 2nd sample was read).

    The 'falling apart' is quite good description as well as there's not enough source samples to be played back so they have to be 'repeated' and if the missing samples have to be calculated (interpolation between values) and the gaps get filled it starts to sound 'muddy'.

    It's different flavors, I prefer the 'falling apart' method for adding dirt.

  • Thinking about it

    there was another old school method to change pitch

    something with messing with the samplerate to change pitch (I forgot how that works)

    I think old ppg stuff did that

  • Some older devices used variable sample-rate to play back samples and then fed them thru a low-pass filter to lower the aliasing.

    With modern systems using a fixed sample rate one way to mimic that behavior is to read the source sample-data at a fixed rate thru a sample-n-hold loop (which sets the sample-rate of the source sample) that feeds the the playback pitch-interpolation with sample values values so each 'sample' used by the pitch-interpolation algorithm is read from the source thru the sample-n-hold loop.

    This way it would be possible to play with the thought of, 'what would this sample have sounded like if it was actually sampled at x frequency and played back at y frequency?'

    A good work-around for now I to link the decimation frequency with the note that is being played back, this way it's per-note decimation which is actually quite nice :)

  • Yes keyfollow to decimator is nice

    it sounds nothing like you would get out of a some down sample plugin :)

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