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5th December 2018, 06:46 PM  #31 
diyAudio Member
Join Date: Jan 2014
Location: Sweden

This might be exactly what i want, if i understand it correctly this is system wide so anything that outputs trough pulseaudio is filtered? Can you add latency on a channel? Is there any latency inherent in the filtering?
One feature that would be nice is if you could import a frequency response and see the changes directly on the curve. Not that necessary or important but would be nice 
5th December 2018, 07:06 PM  #32  
diyAudio Member
Join Date: Jun 2018
Location: Straubing

Quote:
AFAIK there is no latency induced by filtering itself as all the pulseaudio buffers have to be handed around anyways and the simply get put through all LADSPA sinks additionally. Yes, you can add a latency to a path by adding a (sub)sample delay. Not sure what you mean by "import a frequency response" though. Hint: you can show the overall frequency response of the filter chain on an output. 

5th December 2018, 09:04 PM  #33 
diyAudio Member
Join Date: Jan 2014
Location: Sweden

Sounds good! What i meant was to load a measured frequency response as .frd file and see the changes on that kinda like in most passive crossover simulators.

5th December 2018, 09:45 PM  #34 
diyAudio Member
Join Date: Jun 2018
Location: Straubing

OK, will think about that. It probably only makes sense to compare the .frd to the inverse eq curve of a specific parametric eq as you wouldnt want to compare it to adjustments already made and inserted when measuring that frequency response, correct? So then you could adjust the eq so that the combined .frd and inverse eq FR would yield a flat frequency response...

9th December 2018, 03:46 PM  #35  
diyAudio Member
Join Date: Aug 2006

Quote:
Design IIR Butterworth Filters Using 12 Lines of Code  Neil Robertson There's some (quite) easy to read Matlab code in the appendix at the bottom of the page. But the odd order Butterworth filters are generally easiest to think of as the even order BW filter one step lower multiplied by the product of a first order BW filter (5th order = 4th order x 1st order and so on). First order = 1+s Second order = 1+1.414s+s^2 >>> Third order = (1+s)(1+1.414s+s^2) I think that would give a = [1, 2.0286, 1.4762, 0.3714] b = [1, 3, 3, 1] 

10th December 2018, 04:03 PM  #36 
diyAudio Member
Join Date: Jun 2018
Location: Straubing


18th December 2018, 05:25 PM  #37 
diyAudio Member
Join Date: Jun 2018
Location: Straubing

FWIW I still am trying to figure out how to calculate the damn filter coefficients for 3rd order filters. I reverse engineered the code of butter() in scipy.signal, simplified it for the case of N=3 (3rd order) and tried to implement it in C. Here's the preliminary result:
[C] #include <math.h> #include <stdio.h> #include <stdlib.h> #include <string.h>  Pastebin.com What I still need to figure out how to emulate numpy.poly() in this case, which takes the zeroes of a polynomial function and returns the coefficients of it. Any chance sombody knows about this stuff? 
19th December 2018, 12:39 PM  #38 
diyAudio Member
Join Date: Jun 2018
Location: Straubing

Update: somehow I figured out, how to calculate third order filter coefficients and I just released version 1.8 which adds second and third order butterworth low/high pass filters together with a number of small bug fixes and the corresponding ladspat5plugins 1.4 package.
Have fun! 
19th December 2018, 04:53 PM  #39 
diyAudio Member
Join Date: Aug 2008

Hoping for some time over the festive break to try on a Pi..

31st December 2018, 09:02 AM  #40 
diyAudio Member
Join Date: Mar 2018

Respect!
Wonderful project!
Can be extended to a 4 way design? My future project is a 3way active fullrange speaker plus 1 or 2 subs. Thinking to use a 7.1 soundcard with this kind of digital crossover 
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