LP, Fast LP and SV LP filters
-
- KVRAF
- 3369 posts since 16 Jan, 2005 from Ottawa, Ontario
Can somebody please define these. I know how different they sound, I'm just trying to understand the definition better so that I know exactly which one I want quicker, especially the performance of SV Filters. Thanks as usual!
- KVRAF
- 12615 posts since 7 Dec, 2004
"lp, fast lp" this really doesnt mean anything.
a lowpass filter would be any filter which lowers the amplitude of high frequencies while leaving low frequency toward zero unmodified.
the most basic implementation of this is:
buffer = buffer + scale * (input - buffer)
basically, we are scaling the difference between the buffer and the input signal. we are lowering the amplitude of high frequencies, we are limiting the rate of change of the signal which is now represented in the buffer.
here is an example:
input = 0.0
buffer = 1.0
scale = 0.5
first iteration, buffer becomes:
0.5
then, follow along
0.25, 0.125, 0.0625, 0.03125.
the buffer will never reach the input, because we are always reducing the rate of change to a factor of the difference between input and buffer.
the functions for this are:
nroot(n,r) = n^(1/r)
expr(n,r) = exp(ln(ln(r))-ln(ln(n)))
pos(spd,smp) = (1-spd)^smp
spd(pos,smp) = 1-nroot(pos,smp)
smp(spd,pos) = expr(1-spd,pos)
"sv" means state variable. state variable refers merely to the fact you can get different filter modes by performing different calculations on the results of a lowpass.
input - lowpass = highpass, basically
if you took the lowpass of a lowpass, inbetween you would have a "bandpass" responce, therefore:
lowpass1 - lowpass2 = bandpass
if you remove a bandpass from the signal you get a notch taken away:
input - bandpass = notched
etc.
a lowpass filter would be any filter which lowers the amplitude of high frequencies while leaving low frequency toward zero unmodified.
the most basic implementation of this is:
buffer = buffer + scale * (input - buffer)
basically, we are scaling the difference between the buffer and the input signal. we are lowering the amplitude of high frequencies, we are limiting the rate of change of the signal which is now represented in the buffer.
here is an example:
input = 0.0
buffer = 1.0
scale = 0.5
first iteration, buffer becomes:
0.5
then, follow along
0.25, 0.125, 0.0625, 0.03125.
the buffer will never reach the input, because we are always reducing the rate of change to a factor of the difference between input and buffer.
the functions for this are:
nroot(n,r) = n^(1/r)
expr(n,r) = exp(ln(ln(r))-ln(ln(n)))
pos(spd,smp) = (1-spd)^smp
spd(pos,smp) = 1-nroot(pos,smp)
smp(spd,pos) = expr(1-spd,pos)
"sv" means state variable. state variable refers merely to the fact you can get different filter modes by performing different calculations on the results of a lowpass.
input - lowpass = highpass, basically
if you took the lowpass of a lowpass, inbetween you would have a "bandpass" responce, therefore:
lowpass1 - lowpass2 = bandpass
if you remove a bandpass from the signal you get a notch taken away:
input - bandpass = notched
etc.
-
- KVRAF
- 8389 posts since 11 Apr, 2003 from back on the hillside again - but now with a garden!
-
- KVRist
- 333 posts since 2 Sep, 2003 from Brazil
It could if the initial question was FL-related (is it?) LP and Fast LP are among the filter types available in the filter section of the Instrument Channel settings (INS). The other are BP, HP, Notch, LPx2, SVF LP and SVF LPx2. There is also a Fruity plugin called Fast LP. The main difference between LP and Fast LP would be CPU load (and corresponding quality trade-offs).aciddose wrote:"lp, fast lp" this really doesnt mean anything.
-
- KVRAF
- 8389 posts since 11 Apr, 2003 from back on the hillside again - but now with a garden!
-
- KVRAF
- 3723 posts since 17 Apr, 2002 from Scotland
ermmmmmduncanparsons wrote:So when are you going to release anymore plugs, Mr Phut?
DSP
the real answer is - I am working on stuff, but it's very sloooowww. So - sometime

