eduard; a Onepole filter will not

suffice to get rid off all frequencies above the cutoff. Is there a way of having a kind of brickwall filter in chuck?

Oh. dear. I fear we are on the edge of a situation where some massive LiSa abuse to use her as a FIR filter would become almost sensible (loop at -sample rate- over a set of LiSa voices that all hold your impulse responce, writing my_signal.last() to their gain, then starting the sample in non-looped mode. You need at least as many voices as there are samples in your impulse-response. For a high cutoff (and thus short IR) that might actually even work). Fortunately for LiSa there is very little use, IMHO, in emulating high-quality DAC's as those sound like a straight (if filtered) signal by definition. The digital good-stuff (Akai s-612, Roland s-550, Amiga 500, etc) on the other hand sounds like it does due to a rather non-perfect implementation which I suspect involves a plain analogue filter with a huge ripple (which adds character, in retrospect and at second hand prices in a market hardly anyone cares for....) in order to get a steeper roll off that's placed as high (in Hz) as they dared to go, trading fidelity for brightness. I didn't analyse the cerquit but I think it's a good guess that for the S-612 Akai simply bolted a envelope to the same filter and presented that as the modulated filter. So; I'd say add rounding to 8 or 12 bits, try to make the AD and DA stages slightly non-linear in ways that don't perfectly match and tune the filters by ear.... But then again, you may have a very different opinion on what the digital "Good stuff" is, I like the S-612 :¬) Oh, and for bonus trickery; the S-550 has a digital filter that's not quite stable at high cutoffs with modulation and particularly a high Q. I suspect part of the blame lies in it being a 12 bit system and I suspect rounding errors in filter coefficients act up there. This can be charming, at times. Perhaps some of our real filter experts will be able to comment on how likely my theory is, I know it goes semi chaotic in that range, the rounding is my own theory. I believe rounding everything to the sample frequency won't improve matters at such bit-rates either, and I always take it down to 15KHz sampling frequency..... Not sure if that might help or just make it all harder, I'm a bit of a enthusiast for low-fidelity yet nice sounding stuff. Yours, Kas.

Also, to be accurate, should be using a Step instead of an Impulse, right? However, this results in lesser quality sounds...

adc => OnePole lpf => blackhole; Impulse pulse => lpf => Gain g => dac;