FIR crossover for J River or VSThost
an esay way to build a simple FIR crossover PC based.
for j river or other VSThost
using the free software rePhase to generate differents impulses.
rePhase | Free Audio & Video software downloads at SourceForge.net
very steep slopes (till 1000 dB/oct) can be created.
system phase correction
65535 taps IR lenght
VSThost+VSTconvolver(+SIR v1.011 for DRC and phase correction )
asio4all,a virtual audio cable is needed if internal source is used.
HPF filter,sound card output (example )
noise floor is high,laptop internal sound card :(
measure 1M before (3 ways,120 hZ,3500 Hz )
final equalization with rePhase and JBL target curve import in HOLM
to use VSTconvolver:
to make a simple config file
examples of config file
i'll show more accurately how to configure the chain
It looks great!
Can you (or somebody else) please write a step-by-step tutorial?
I tried setting it up before but I wasn't able to get it working.
By the way is non-linear phase audible to you?
What does it sound like?
yes,i'm gonna give more details
for my little ears,i do not found difference between linear phase and minimum phase,but it's free to do,so let's do it !
i've spend some time to make it working but now it's very simple to build
today,it's late in France.:sleep:
Enjoy your rest!
You deserved it :D
I've downloaded it, and look forward to playing with it. Thanks.
The phase thing in general is very interesting, and something I don't yet feel I entirely understand.
In the second illustration down, the phase goes through several 'wraps' of 360 degrees. This might be explained by a constant delay caused by, say, the distance between driver and mic, with the true phase shift of the speaker riding up and down on top of the linearly descending line, presumably(?). If we somehow infer what the constant delay is and remove it, and yet the phase still goes through several 'wraps', can we still linearise it? Am I really talking about 'minimum phase' vs. 'excess phase'? (two terms whose meanings I don't think I fully understand yet). Programs like REW allow you to export what they call a minimum phase version of the impulse response. How do they separate the minimum from excess phase, and does your program implicitly do the same thing, Thierry?
yes,we can linearize all phase shift,with fine work,phase can be limited to +-20° on the 20 Hz-20 Khz band.
this is not my software,my knowledge do not allow this powerfull tool
i hope the author comes to give some clear explanation.:)
the second screenshot shows the phase shift of the loudspeaker at 1M
this is the minimum phase.(combination of electrical phase shift+acoustic phase shift ).
micro is calibrated spl+phase
the measurement shows,in this case (3 ways )
for helping,to correct phase shift,detect the 0° of the phase shift (in reality this is multiple of 360°)
-35 Hz bass reflex tunning
-120 Hz analog active crossover
-3500 Hz passive crossover
with rePhse,the idea is to create a inverse phase shift,generate an IR,apply it to the signal before amp...and it's working...at 1 M.
at the listening point,reflexions of room cause more severals phases shift.
-at 1 M phase shift is about 550°
-listening point,phase shift if above 5000°
i'll show how to use,easily,rePhase to get a loudspeaker linear phase.
1- use the filters linearization tab
insert the theoritical phase shift
-bass reflex or closed box
2- use the PPEQ to refine the curve
to create FIR crossover
see the links before this post
click on the tool tab for advanced config
click on blue tab to active input and output as shown below
configure asio input and output selection
configure output engine and input engine
it's up to eachother to choose wich channel gonna be used
i'm using 1+2 for LPF (front left+front right )
5+6 for HPF (rear left+rear right)
a sound card can be added to reach 2x8 channel,for tri or more multiway systems.
this setting takes less than 2 minutes once driver installed.
now,VST host is ready to work
next step generate IR from rephase
and insert IR in the convolver
later,i'll explain how to easily add a second convolution for DRC+phase correction.
tab linear filter
create an Impulse response you need (kind of filter,frequency,slopes)
adjust the graphic Y axis at -150 dB.(to see eventual ripple)
choose the same number of taps for each IR.
low pass IR (+bass boost if you need with GEQ section )
and another one HPF IR
open wordpad and write:
save with a name you're choosing
explanation (same on VST convolver site)
and select the file .txt you've writed
configure all your media player output--->virtual cable
same with HOLM,REW or ARTA
you can check the result with a cable looping back to line in
use multiple instance of VST convolver is difficult.
for DRC and phase correction,
use SIR V1.011 freeware to insert a second (or more ) convolution
i'll will explain the way to do it
it's important to use the same number of taps for each IR in the cross over,delay will be different otherwise
to reduce delay of processing,increase the partition (paralelling process ) less delay,more load processor
it seems to be heavy,but it takes less than 10 minutes to configure
after,let's play with rePhase and differents setting of crossover
( a DX46 FIR commercial crossover cost 2000$,with a max of 512 taps ).
with a basic PC,you get a powerfull Xover+DRC,a full phase linear system,without external hardware.
only power amps connected to sound card.
i've try on a basic laptop,10 stereo convolutions (8000 taps )
load processor is about 23 %
with the desktop,the final config:
1 stereo convolution for DRC and phase correction (8000 taps )
2 stereo convolution for crossover 2 ways (8000 taps )
processor load 5%
example for 3 ways stereo config
10 ms delay on HPF filter
-6 dB on the bandpas filter (coeff .5 )
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