PlanetZ forums - scopeusers.com

Hi,

Here's a little test of 2 different lowpass filter designs - which one is the best implementation regarding low frequency response etc.? Please modulate the parameters all you can.

Unfortunately the C1 & C2 are not calibrated - so they wont match - that will of course make the test abit more difficult.

C1 = C2 = Range [0 ; MAX]

http://www.relab.dk/Filter.rar (new upload - should be easier to follow)

Cheers

Thanks Martin.

Im not sure for what

I think this is only usable in Scope SDK - It's not a device, but only a test of different filter topologies - sorry.

Well I just found that out.
I tried to load it as an insert in Prowave................No Bueno.
So I then loaded it into the project window and ran my analog synth through it, at a very degraded signal.
I should have known when it had no GUI that something was wrong.

Anyway Thanks for building and developing for Scope SDK then.

Happy New Year.

New upload - should be easier to follow.

noone responding here?

is there a reason for sync controles?
do you want to modulate?
---------

for low frequency these are about the same to my ears;
havent checked if theres a difference in circles;


LP2

i wouldnt use for audio;
but it has its existance for analysing;
thatfor i would use async singal for controlling to save a lot of circles;


LP1

sounds better to my ears in the high frequency range;
only probl. is the recursive tap itself:
at max should be a cutoff Samplerate/2 but is not! its a bit lower;
this is like the stock atoms filters and was the reason for my personalstudy;
youre able to solve this by lowering C2 from last add (mix3) when the max controll is reached;
but im sure you know!!

so to your question:
LP1 is better the higher the cutoff is;
with the probl. of the last few HZ up to SR/2 where i would prever LP2 aproach;

good vibes from vienna

Hi tgstgs,

Thats correct - the sync module is for modulation purpose.

I have uploaded yet another version - now including the Chamberlin form (LP3) - http://www.relab.dk/Filter.rar

The LP1 form is a low sensitive structure - better response than LP2 with modulation.

Not many are apparently interested in filter structures

had lots of fun with matlab;
check it out;

porting c to sdk i was not able to beat stock atoms circles;
need to build in assembler;

im sure with xite a lot of high order exotic filters are possible
time will tell . .
----------


will test lp3 later

good vibes

hi all,

Not many are apparently interested in filter structures

noone responding here?

actually an intereting subject ... i follow it as possible ... and maybe will post , but miss time,

let me just say that DSP Equing/filtering is a wonderfull world, full of maths and passionating things (truncations, approximations, nodes overloads ...)

we at DAS have spent 6 months to realise our master it eq algo,

Martin thx again for your contribution to our knowledge

cheerz
olive

Hi Martin, we're just back from spending xmas with family. I'll check out your filters as soon as I can get access to the sdk, but I'm not quite sure when that will be...

cheers,
Mark

hi martin,
i had a look this afternoon at your designs , actually interesting , espacially the LP3, many possibilities as "musical" filter ...

there are some things i don't understand ... Mix 3 instead of Mix 2 (that would make the dev lighter ... (dsp consuption speaking, ) but i guess you have something else in mind !

LP3, "sounds" great ... i didn't know this implementation, quite versatile and really "optimisable" ...

in term of "strict low pass", i would prefer LP2, but at this time i must admit that i didn't passed audio throu it, only "fft analisys" ... and i do know by experience that it is not a good judge ...

i am still really interesed in your further researches ...

cheerz
olive

Sorry the Mix3 modules could easily be replaced with Mix2 - they were used to correct the summing issue of positive numbers - but that should be handle differently because of negative numbers.

The filter structures are without saturation and randomization - that should be implemented somewhere in the feedback loops for different effects.

I'll wait before I release additional structures - some could also cascade 4 LP1 with feedbacks - ladder type filters.

Oh and the sync modules is only for audiorate modulation.

Cheers

sorry for late reply

last download here in topic LP3 is oszilating;
= not stabil;

im hear a fu* ringing with low input at about -12dB _ NO???


good vibes

tgstgs wrote:LP3 is oszilating;

Only at MAX res

have to agree with olive LP3 sounds very good;
i tested with a guitar doing sort of wah effect with a controller;
specialy on low frequency input i like what it does to the higher parts of the guitar spectrum;
----
this is definitive worth further development;
have to test its switching characteristics;
need some time

good vibes

I am very interested in taking a look into those structures, but didn't find the time so far. I bet I missed something, I guess I will not understand anything though . Thanks anyway, I'll comment when I find the time for testing.

1000 Thanks, exspecially for your old topic on reverbs too!!! It's so cool to have people here sharing knowledge on such a high level.

Martin II

Well - after this success and awesome participation its about time to continue.

I hope most people looking at the structure found out how to extract highpass and bandpass simultaneously from Filter3 - otherwise you can download it here : www.relab.dk/Filters.mdl

If you sum the LP3 and the HP3 you get band reject.

Thanks, I'd like to get involved in this discussion when I've got some free time with the sdk again.

warp, do you know how you can build oversampling filters via sdk?

i ve not yet found a way, surely by touching the assembly code, but with the basic atoms?

Not by basic SDK atoms.

I do have access to STW oversampling modules.

I talked with Jurgen at the Musik Messe regarding oversampling modules for the XITE system - Sonic Core will look into it - the modules will have additional in/outputs for oversampling - 2 in/outputs for 2x oversampling and 4 in/outputs for 4x oversampling.