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Understanding the State of the Art of Digital Room Correction

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It's wrong because it introduces pre-ringing and that is audible. With minimum phase FIR you don't have pre-ringing.

Hopefully this picture will help explain:
Thanks for trying to help but my question was to JJ in relation to the comments he made. I am aware of the difference between them and to say that all pre-ringing is audible isn't supported by any studies I have read.
 
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about multiple meassurements:

I have tested it once. a single meassurement X in var smoothing equals to averaged symetric multiple meassurements around x. of course this will be the case in a symetric room only.

all that smoothing stuff doesn't make any sense to me though.
the only smoothing that would make sense is to ignore frequencies between notes since they wont be active with western music. though even in western music you will have overtones outside of the scales
 
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Thanks for trying to help but my question was to JJ in relation to the comments he made. I am aware of the difference between them and to say that all pre-ringing is audible isn't supported by any studies I have read.

Well, you asked "why is a symmetric FIR wrong", which is strange if you are really aware of the differences between them.

I never said all pre-ringing is audible, but pre-ringing of symmetrical FIR is pretty much always audible. With minimum phase it practically never is because of masking. In pracice, well designed room correction FIR filters are not exactly minimum phase but are pretty close to it, so while you will notice some mild pre-ringing on their step response it will be inaudible.
 
Well, you asked "why is a symmetric FIR wrong", which is strange if you are really aware of the differences between them.
Maybe it is but earlier I said
Most FIR filters use a centred impulse alignment where the latency due to the filter alone is half the sample time of the overall filter as shown above. Linear phase filters have symmetrical impulse responses so this makes sense. An acausal filter could have the impulse placed anywhere within the filter but that would limit the amount of time correction to the number of samples of delay so is more unusual.
On reflection I should not have said most or more unusual. You even liked this post :)
I never said all pre-ringing is audible, but pre-ringing of symmetrical FIR is pretty much always audible.
You may have meant that but you said
It's wrong because it introduces pre-ringing and that is audible.
Which isn't quite the same and prompted my reply.
 
the problem is that you are asuming that "post ringing" is inaudible. it is audible:

In modal equalization the idea is to match ringing of mode and filter so they perfectly cancel each other.
 
Not at all but all I've read so far didn't really answer the questions I had and still have. Maybe you have better applicable references? For what it's worth I went through a ton of papers (including yours) and Jan Schnupp's book.
Well, next step is to consider the shortest impulse response one could get knowing the bandwidth of the various cochlear filters.
 
I have a hard time understanding the use of critical bands for frequency response of a reproducing system.

let's take a FR, it is flat on the erb scale:


View attachment 161936

in "reality" though 55Hz (A1) is 3.6dB louder than 52Hz (G♯1):

View attachment 161937

If I play a "melody" alternating G♯1 and A1, I wont hear a diference before and after using this filter?
I would suggest that rather than spread confusion, you go to www.aes.org/sections/pnw and look in meeting reports for "Hearing 099".

Hint: There's not ONE filter per erb, not by a long shot. On top of that you're dismissing phase response above and below the center of the filter, and a bunch of other stuff. Since you seem annoying and insulting, and appear to be more concerned with winning points and muddying the issues, I think I'll let you figure it out.
 
In modal equalization the idea is to match ringing of mode and filter so they perfectly cancel each other.

the main thing mentioned DRC software adresses in the time domain is the group delay in the bass region. in my video I showed a typical group delay of about 10ms afair
 
Thanks for trying to help but my question was to JJ in relation to the comments he made. I am aware of the difference between them and to say that all pre-ringing is audible isn't supported by any studies I have read.

Ah, yes, the "straw man" rhetorical cheat. You're the only one to use "all". "All filters that are long enough to go a good job" is a different story from "all".

Obviously a 3 tap filter of .05 .5 .05 is not going to have audible pre-ringing.

For future reference, avoid these kinds of things, m'kay?

 
What?? how is that? I just asked a question. I will try to understand the tecnical papers, but about my question I will probably let my ear decide

I will test this filter instead. it is almost flat on the most used VAR smoothing and in no smoothing has over 10dB diference:

c.jpg


if I hear no diference beween the notes I will apologize, if I do I will probably not come back to the topic since argumentum ad verecundiam is so strong in this forum. if anybody is intrested in a AB topic let me know though
 
Uh, for this, using ambisonics (not my favorite tool, but for this it's good) or related techniques actually helps with that.


In particular, by capturing the volume velocities in a room, we can very nicely recognize the stored energy in a room.

"Stored energy". Isn't "stored energy" the basis for any "active absorption" approach? Why do you consider this an issue? A DBA that delivers near perfect results is "not a good solution"?
 
"Stored energy". Isn't "stored energy" the basis for any "active absorption" approach? Why do you consider this an issue? A DBA that delivers near perfect results is "not a good solution"?

Active absorption? Really? Now that's unique. You seem determined to pick an argument, but I have exactly zero idea why you're arguing about this with me.

As to "stored energy', yes, knowing the stored energy is the absolute key to knowing how to deal with low frequencies without requiring multiple, accurate mic placements.

And by knowing all 4 parts of the soundfield at the central point, yes, you can do a very accurate job of that. Now, by four points I mean dx, dy, dz, and P.
 
I will test this filter instead. it is almost flat on the most used VAR smoothing and in no smoothing has over 10dB diference:

View attachment 161947

if I hear no diference beween the notes I will apologize, if I do I will probably not come back to the topic since argumentum ad verecundiam is so strong in this forum. if anybody is intrested in a AB topic let me know though

What in creation are you talking about? You're oversimplified things to a degree such that I don't go where to start.
 
Well, next step is to consider the shortest impulse response one could get knowing the bandwidth of the various cochlear filters.

Sorry doesn't compute. Could you please elaborate? As I understand it "cochlear filters" are just a model to explain certain mechanisms of our hearing but I fail to see how it is directly applicable to creating room correction filters. I tried but failed.
 
Active absorption? Really? Now that's unique. You seem determined to pick an argument, but I have exactly zero idea why you're arguing about this with me.

As to "stored energy', yes, knowing the stored energy is the absolute key to knowing how to deal with low frequencies without requiring multiple, accurate mic placements.

And by knowing all 4 parts of the soundfield at the central point, yes, you can do a very accurate job of that. Now, by four points I mean dx, dy, dz, and P.

I'm not arguing, I'm asking. Your answer isn't helpful to me. Are there aspects you're not free to talk about (which is perfectly fine) or are you just unwilling?

Haven't read that paper (yet) but in other papers you suggest the use of a microphone array? This would be totally different from what mitchco does.
 
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