wow some really good info here in this topic… <3
Bring on the filter mod 
OK, this is going to be NUTS %) x6xx here I come…
<< letz juuse up da basslinez bitchez :-o explodes >>
Allow me my quota of one immature teen moment won’t you 
I was working on a quite big song which has been started from the template of another. as soon as I was sure that some tracks of the old song were unnecessary, I deleted them. Some minutes after, having changed a lot in a pattern, I realized that I accidentally deleted track12 aswell, which indeed had to be kept!
so I copied the modified pattern, went back with the undo until track12 reappeared, pasted the copied pattern, which did not have the deleted track, and everything worked fine: the result of pasting was the cchanged pattern, with a blank track12, and of course all the rest of the song had track12 content intact!
do you remember those FT2 days when you didn’t have undo at all?
I’m with you on that Itty. Many many times now in 1.9 I’ve totally relied on the undo going back back back back back… and everything works ok.
Just looking at the new version now with the ‘corrected’ filter titles. I’ve grown so use to the old titles with the references to the slope. I was understand, perhaps incorrectly, that the slope meant what it did (although the cutoff in some filters was confusing). Please can we get some clarification on that, as well as some explanation behind the ‘pole’ terminology. I like to learn 
In the meantime I’ll do some of my own research and post back here.
http://www.soundonsound.com/sos/sep99/arti…ynthsecrets.htm
If you’ve been in the synth game for a while, you’ll have heard that 12dB/octave filters are sometimes called ‘2-pole’ filters, and 24dB/octave filters are called ‘4-pole’ filters. You might think it safe to assume, therefore, that each of the 6dB/octave sections in Figure 9 is a ‘pole’. Unfortunately, you would be wrong (although not a million miles from the truth).
The name (in this context) is a consequence of a powerful mathematical operation called a ‘Laplace Transform’. This transform, while difficult to describe in words, is a convenient operation that allows mathematicians to analyse the responses of linear systems when they are presented with audio signals (as for ‘linear systems’ and the maths involved… no, don’t even dream of asking!) Anyway, the term ‘pole’ comes about because, when you represent an RC filter using a graph in the ‘Laplace Transform domain’, it looks like a flat sheet of rubber with a tent-pole pushing it sharply upwards at some point. A single 6dB/octave RC filter has one such ‘tent-pole’, and is therefore called a ‘1-
“Filters with a 6dB/octave characteristic are used as tone controls in stereo systems, and occasionally within synthesizers as supplementary brightness controls, but they are not much use for true synthesis.”
pole’ filter, a 12dB/octave filter has two ‘poles’… and so on. Therefore, if you want to create a passive 24dB/octave filter with a single cutoff frequency for each of its four elements, it would seem safe to assume that would you want all the poles in the same place in the graph. And, for once, intuition is correct. Unfortunately, as I’ve already explained, achieving this using passive components is all but impossible because, when we cascade the sections, they interact and no longer function as they would in isolation. So, instead of the perfect 24dB/octave response of figure 10, the cutoff frequency for each section is different, and the amplitude response of our transfer function has four ‘knees’, as shown in Figure 11.
This then, leads us to an important conclusion: while a passive 4-pole filter will tend to a 24dB/octave rolloff at high frequencies, it will, to a greater or lesser extent, exhibit regions within which the rolloff is 6dB/octave, 12dB/octave and 18dB/octave. Moreover, if you look closely, you’ll see that the transfer functions within these intermediate regions are not quite straight lines, meaning that the relationship between the frequency and the input and output powers are not as straightforward as before.
Ok so now that I understand this, I’ve one remaining Q: What is the slope of the Moog filter?
Yeah that’s great, certainly.
But for more useful visual undo control - undo list would be much appreciated!
Indeed. Maybe for 2.0?
woah… did we not have undo in ft2? =)
hmm… please confirm someone =) I’m interested
oldschool trackers never make mistakes.
^^^^^
lol!
Yeah, I noticed more CPU usage too. With the RC1 it seems to be much better. Have you tried this version yet, and if so, do you still have issues?
Yeah…
=)
Anyways thnx It Alien… Didn’t remember this. Having undo at hand can be quite limiting and I think I’ll go back to the ft2 style… =)
word, oldschool trackers save a lot 
I have a song that only works (or rather, glitches as it should) in Renoise 1.52, so I still have those days whenever I edit that song 
(it’s amazing how far Renoise came since then, the difference between 1.52 and 1.9 is just sick!)
if I’m correct, it has always been like this, because one could want to apply an LFO to the volume even if it has no envelope.
Yes, exactly. I see no reason why they have to be linked…
this leads to an old question of mine: wouldn’t be great if we could use the LFO as a relative oscillator? The current one is always an absolute LFO, which is kind of unusable with fading envelopes because the oscillation is not related to the current envelope amplitude.
just to resemble the old question:
imagine you have an envelope ramping from 100% to 0% volume, with a 25% LFO.
at first, the LFO modulates the signal between 87.5% and 100% and then, as soon as the envelope starts to fade out, it goes through 75-100%, then 50-75%, 25-50%, 0-25%… then… 0-12.5%… and never stops!
a relative LFO is in my opinion something really useful.