from what i experienced if i resample on the zoom st224 many times , the sample will add a little "space" of something between it, and on the sp303 too...it causes a sort of delay and i think this is like a sort of sum "something" to the original sample...i noticed this on softwares also (not the delay, but some "air" to the volume)

Actually, before we both make ourselves look even more ridiculous here ...I think we're perhaps arguing the very same thing here as I've never mentioned converters in the process, but talked about a separate sampling engine.

I'll admit that's probably much closer to digital than I was stating/assuming.

However.. it really can be done in both ways actually! Analog processors operate directly on the electrical signal, while digital processors operate mathematically on the binary representation of that signal.

But a chip is still a chip if you catch my drift here.

No we're not talking of the same thing. When you say "resampling the analogue way", the signal MUST run through the DAC first, otherwise it would still be digital. That also means of course, for sampling that again, it would need to run through the ADC.
Today, a sampler is an entirely digital device. If you would leave out the converters and only have digital inputs and outputs, you wouldn't have a single bit of analogue audio circuitry in there. There HAVE been samplers in the past with analogue filters, but they're an exception. By chance, I even have one, teh Prophet 2000

And no, "a chip is not a chip".
For example, there are people mistaking every chip as a digital device. As you surely know, that's bullshit. An IC (proper term) is nothing more than any kind of circuit miniaturised and put into a (mostly plastic) carrier. The circuit inside can be of any kind you want, analogue, digital, or mixed (hybrid). A classic example for hybrid ICs are DACs and ADCs, as they contain both digital and analog audio circuitry.

But I agree, we shouldn't continue here because it surely turns into ridiculousness. Just try getting your facts straight on what is analog and what is digital audio circuitry. That's my last call.

You've perhaps missed something I've mentioned though;

"However.. it really can be done in both ways actually! Analog processors operate directly on the electrical signal, while digital processors operate mathematically on the binary representation of that signal."

They're still just processors, only the processing is different.



Filters... you mean for envelopes, modulation or layering? Then you're talking about something different then I am.

I didn't miss on that, it just has nothing to do with the question whether resampling is done digitally or not. Sure both ways are suitable for processing audio, generally spoken. But then, lions eat meat and humans do too, if you excuse the bad comparison


I mean filters for filtering the samples, like resonant lowpass/highpass or even bandpass, you know, synth filters used for sound design. Anyways it was just an example of exceptional analogue tech inside an otherwise completely digital system.

I'm sorry but that usually is modulation, envelopes and layering. LFO's etc... the same stuff common on analogue synths indeed.

That too can be done both entirely digital or with analogue filters though.

I'm not sure what point you're trying to make here, as I do not disagree with any of that.

No, filters don't modulate, they filter as in removing certain frequencies from the sound. Parameters of the filters like cutoff frequency or resonance are getting modulated by modulation sources like envelopes, LFOs etc.
But that's besides the point.

Look again at the drawing I made. Every signal below the converters (the A/D blocks) is digital, everything above is analogue.

Is it so hard to understand that establishing the resampling loopback in a way it goes through both converters makes no sense, neither from the engineering/cost cutting nor from the sound quality point of view, and that this is what makes it highly unlikely that any manufacturer implemented it that way? The engineers at Roland or E-MU or AKAI are not stupid.

the A<B<C<D banks are a joke......i bouhgt tha kp3 recently and the internal clock is off...that means that i record a loop at A, resampling it at B , and after a while they are off ...cant play live like this....
total crap!

this is written in korg forums for kp3...
can keep time with itself but drifts out of time with external gear when playing back samples....


does this happen to Kaoss pro as well????

Wasn't that a bug of some kind in the KP3?