Except that this time the room is involved. Clearly you haven't tried any of this or you'd know that. If you work on it, you can make it work out better than using a regular crossover created the anechoic way.
Not that you should or have to try it though, but just don't judge it this easily or wave it off like this. Be happy with what you've got and we'll do the same.
I hope that anyone interested in this thread, is at the level where they take all you said as a simple set of givens.
Not sure why there is even a need to mention that there isn't any competition between quasi anechoic and room based measurements.....
Simply different measurements for different purposes.
Personally, my purpose is I'm still not interesting in room/system optimizations.
I experiment with too many different speakers, and in too many different rooms/environments.
In my usual main testing room, speaker and sub placements are constantly moving around; as is whether I'm using a mono, stereo, of 3 ch LCR rig.
The idea of taking speaker measurements within 1/4" distance accuracy, per stereo room tining recommendation, is a complete non-starter for me.
And I enjoy real-time tone controls. I don't begin to believe "a set it and leave it" house curve/tuning can work given tracks' tonality differences. Particularly with their variances in bass.
I enjoy real-time tone control listening to my Stax headphones, and find I use close to the same tonal adjustments for a track with the phones, as I do with speakers in whatever setup I'm running in whatever room.. So I impute my preference for real-time tone control is not about overcoming room tuning deficiencies, but from the tracks themselves.
I'm still learning new things about quasi-anechoic measurements and tunings, and want to feel I've exhausted that route before moving onto (what I see as more advanced) room system optimization.
I say all this simply to let folks know I'm not at all advocating that quasi-anechoic tuning is a final end-all approach to tuning.
Although I do advocate that getting it as correct as you can, can't help but yield better in-room results, when room optimization is then added on....
which is just common sense i think.....
At lower the lower frequencies we talk about there may be more suitable solutions (if that's your goal), based on placement and inherited room effects.
(multi sub starts to make a lot of sense)
One can create the perfect speaker, but place it in a lesser room, and it won't live up to its potential.
I think ultimately when one is trying to optimize a particular speaker/room setup, that has to be true.
Although I find it a bit ironic, that a speaker is at its best when first optimized independently (anechoic),
and that the same is probably true for a room,... best if first optimized independently acoustically..
Yeah, I know a lot of folks swear by multi-sub....I just don't like the sound of it.
I find bass transients from multiples often don't have the intensity...particularly tactile.
^ rather than the sound of it i've been more bothered by the implementation of it ( in the Geddes's approach).
It can present some challenges about real life implementation too ( cables running through the room, number of out required).
If the goal is to have more coherency over a large area multiple subs definetly have an edge in my view.
If it's not a primary goal then other approach seems to work too ( seems as i have not implemented the one i think about seriously so can't comment on it).
It can present some challenges about real life implementation too ( cables running through the room, number of out required).
If the goal is to have more coherency over a large area multiple subs definetly have an edge in my view.
If it's not a primary goal then other approach seems to work too ( seems as i have not implemented the one i think about seriously so can't comment on it).
I mentioned a lesser room for a reason. You'd be absolutely right if one had the opportunity to build a room suitable for the speakers.
We don't always have that luxury. Some speaker designs help in that regard, the Nao-Note from john k... certainly falls into that category. Yet, without turning every room into an ideal acoustic environment, one can still find steps to advance the results of the room + speaker. Especially if one starts with a good speaker concept and a maybe not so optimal room to put it in. One can optimise the results by using the characteristics of both to your advantages. And that doesn't have to include the idealized crossovers that would work well in an anechoic space. The mains could be part of the multi sub solution if they have that low end potential and it could spare you from needing as many subs (and cables across the room). Letting go of the perfect ideal and being creative about it in a lesser space can get you ahead. The world isn't black and white.
If you "get" why it works, placement can bring you much more practical solutions. Especially if you can control each part that makes up the low end separately.
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They work , just not very good…
And it takes too much to get it to work the way I think we want…
There’s got to be another way to average the data and keep the phase data…. Like some math stuff I’m sure. I just a dummy at math.
The db average response is what I would love to get with phase data attached…
Idk maybe some trace math to get it to do what I want…. I’m just not smart enough to do it I guess…. (Me and about 99% of the tuning population because you guys as much as I admire this group your all just way smarter then me… )
I'm sure that can, and most often does, get you ahead.
I've tried to explain why I'm currently not very interested in trying to optimize a speaker system to a room.
I hope you can understand that I view the potential gains from room/speaker optimization as small,
compared to the potential gains I've been getting from continued DIY speaker work.
Too small to fool with at this point in time....still after bigger fish....
That said, if my latest large full-range synergy that incorporates 18" drivers doesn't pan out...I may have hit a point of no more ideas for significant DIY improvements. (Thought I was at that point with the 3 syn LCR setup, but ye ole light bulb went off)
Then it might be time for some room opt work. (But maybe not....I can't tell you how happy I am with what I already have, ....as is....)
A pure aside, the biggest issue I see going forward with DIY syns is the inability to get baltic-birch....the Ukraine war has totally dried it up here in the states.
My speaker building may be at and end, just do to that.... because even if I love the new big syn, i can't build another until BB supply resumes, or a substitute is found...
I think it's a global shortage Mark, in here finding some BB is a pita and ask for BIG money.
I tried to ask for bamboo ply and got a strange look from vendor... i think he didn't know it exist! 😱
I've proposed Okoume ply. I don't know the material and given it was not cheap i dropped the idea for now...
Anyway as i contemplate something needing a lot of prototype i'm thinking foamcore atm... building a hot string table!
I tried to ask for bamboo ply and got a strange look from vendor... i think he didn't know it exist! 😱
I've proposed Okoume ply. I don't know the material and given it was not cheap i dropped the idea for now...
Anyway as i contemplate something needing a lot of prototype i'm thinking foamcore atm... building a hot string table!
Ok, take it or leave it. I did this this afternoon in about 30 minutes one I got all the stuff set up. All measurements were taken at the same position.
1) I measured the response of my systems woofer at the listeng position. Pretty nasty.
2) I defined a target, B3, 30 Hz HP, LR4, 125 Hz, LP.
3) I generated the minimum phase EQ.
4) I started the play back of the Eqed woofer and measured it (had to use my old measurement system since the other PC was doing the filter processing). Not perfect, but 15 minutes work. And I could easily take this and generate a 2nd correction filter to improve the result.
Now, if you want to mess withthe phase it's a separate issue.
1) I measured the response of my systems woofer at the listeng position. Pretty nasty.
2) I defined a target, B3, 30 Hz HP, LR4, 125 Hz, LP.
3) I generated the minimum phase EQ.
4) I started the play back of the Eqed woofer and measured it (had to use my old measurement system since the other PC was doing the filter processing). Not perfect, but 15 minutes work. And I could easily take this and generate a 2nd correction filter to improve the result.
Now, if you want to mess withthe phase it's a separate issue.
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Use a frequency dependent window for starters. It's all there if you really care enough to read it.
Everyone knows how to make a few tweaks with PEQ. That's not or never has been the point we made.
One of the things that make it more complicated is to be able to find the right frequency dependent window.
A simple on axis measurement isn't going to cut it, sorry.
You don't have to try though, wouldn't want to steal that much of your time to catch up. Just don't belittle the time we spend to figure it out.
The example you made in about 30 minutes screams you didn't get it though. We said it would be more complicated than that, didn't we?
I do realize you're kind of a hero around here and all for good reasons, but maybe, just maybe the time we spend to figure out this stuff wasn't all just hot air.
Everyone knows how to make a few tweaks with PEQ. That's not or never has been the point we made.
One of the things that make it more complicated is to be able to find the right frequency dependent window.
A simple on axis measurement isn't going to cut it, sorry.
You don't have to try though, wouldn't want to steal that much of your time to catch up. Just don't belittle the time we spend to figure it out.
The example you made in about 30 minutes screams you didn't get it though. We said it would be more complicated than that, didn't we?
I do realize you're kind of a hero around here and all for good reasons, but maybe, just maybe the time we spend to figure out this stuff wasn't all just hot air.
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No need for sarcasm. I still haven't heard a reply as to what frequency dependent windowing gets you or what it really does. Just "you have to get it right". Tried to research it but nothing shows up in a google search. I'd love to be pointed to a good explanation. And I really haven't read a definition of what the desired result is. Perhaps if the problem were clearly stated I could actually help. Still my approach is that what every the problem is, you have to start by solving it at a point.
A few pages ago Fluid (I think) said he wanted a system that was minimum phase HP at the listening position. To me that implies two things: 1) correcting for crossover induced phase and 2) correcting for phase nonlinearies introduced by reflections. The first is straight forward. The second, not so much, at least at higher frequencies.
A few pages ago Fluid (I think) said he wanted a system that was minimum phase HP at the listening position. To me that implies two things: 1) correcting for crossover induced phase and 2) correcting for phase nonlinearies introduced by reflections. The first is straight forward. The second, not so much, at least at higher frequencies.
You weren't meant to try and read the values, I have said I cannot give you a value as it varies. I could tell you what I use but that is for my speaker in my room, yours will not be the same so it has no value to you.
The image was meant to be illustrative to show the relative lengths of the windows with different exponents.
Absolutely this is subjective, the trick is to be able to correlate an objective measurement with a subjective response. Finding the right measurement to base the correction on is the fundamental step in this process. For myself I have found that there is a point where things move from good to excellent to right. Right is when there is nothing about the sound that I want to change, no fault I can identify. For windowing the sweet spot is where there is maximum improvement in subjective sound without any artefact from the processing.
Frequency dependent windowing allows the strength of the correction to be varied by frequency
You will find almost no formal direct research on the topic.
The desired result is that the listener prefers the sound of the system more with the correction in place than without it.
My actual point is more nuanced than that, but it is very hard to make it without confusion. I think phase matters but mainly at low frequencies below 1K and more importantly below 400Hz. So if the crossover is above 400Hz linearizing it might well be very hard to hear. What I don't want to do is linearize the boxes natural high pass response, I do want 2) in your list at low frequencies but with a subtle touch not a brute force assault.
I guess this wiki from Audiovero helps: https://www.audiovero.de/acourate-w...ionen:td-functions:frequency_dependent_window
Watching the video that Mitch posted a page ago: https://www.diyaudio.com/community/...earization-by-fir.393435/page-32#post-7313962 might have solved that part too.
I'll include it:
To be able to determine what to correct and especially when to correct it, one would need to examine the effect of the room on the speaker. Not only at the listening spot, but also at various points of interest around it, as to be sure that the correction won't be a "one spot" solution. That sure is not as easy as just making a few tweaks with PEQ. The above mentioned frequency dependent window is a tool one can use to look at the "early" wavefront. At higher frequencies, you'll be able to go in, see the early output before the reflections have happened (much like a properly gated window) and if needed, base your EQ on that.
At lower frequencies, the room effects will be part of that early window, no matter what one does. But programs like DRC-FIR (a free tool), Audiolense and Acourate (from Audiovero) let you vary the frequency dependent window used for magnitude correction EQ and even use a different sized window for phase EQ(*). Next to this handy feature, they usually have a zillion other options that determines how the program needs to behave. In DRC-FIR (the free program) you can set just about any (of those zillion) variable(s) to your liking, but that also means a lot of legwork from it's user to determine what does what.
The Acourate and Audiolense products follow a more pré-determined path and are more automated, based on the choices of their respective makers. There are small differences between these programs, as far as the inner workings go, but that does not detract from their purpose.
Personally, I've studied my speaker in my room and what I could do with a tool like the above mentioned solutions. Mitch has been a big inspiration in my early days of figuring this stuff out and we've compared notes on several occasions. Same goes for fluid and I, and many many more users of these tools.
These tools come in handy but require work to get a meaningful result. It would take me a long time to go over it and explain all they do, but I won't in this thread. I do have a thread which is almost 400 pages long that has a lot of this stuff covered, open for anyone to read and you'd be surprised how many have already done that.
There's a thread on the full range side of this forum that has helped a number of people get started with DRC-FIR. But it's still just a tool and any tool is as good or as effective as the user that determines what to do with it. It does tend to take time though, as we tried to explain earlier.
(*) you can even limit the phase EQ to a certain bandpass, for instance to only use it on that low end.
Watching the video that Mitch posted a page ago: https://www.diyaudio.com/community/...earization-by-fir.393435/page-32#post-7313962 might have solved that part too.
I'll include it:
To be able to determine what to correct and especially when to correct it, one would need to examine the effect of the room on the speaker. Not only at the listening spot, but also at various points of interest around it, as to be sure that the correction won't be a "one spot" solution. That sure is not as easy as just making a few tweaks with PEQ. The above mentioned frequency dependent window is a tool one can use to look at the "early" wavefront. At higher frequencies, you'll be able to go in, see the early output before the reflections have happened (much like a properly gated window) and if needed, base your EQ on that.
At lower frequencies, the room effects will be part of that early window, no matter what one does. But programs like DRC-FIR (a free tool), Audiolense and Acourate (from Audiovero) let you vary the frequency dependent window used for magnitude correction EQ and even use a different sized window for phase EQ(*). Next to this handy feature, they usually have a zillion other options that determines how the program needs to behave. In DRC-FIR (the free program) you can set just about any (of those zillion) variable(s) to your liking, but that also means a lot of legwork from it's user to determine what does what.
The Acourate and Audiolense products follow a more pré-determined path and are more automated, based on the choices of their respective makers. There are small differences between these programs, as far as the inner workings go, but that does not detract from their purpose.
Personally, I've studied my speaker in my room and what I could do with a tool like the above mentioned solutions. Mitch has been a big inspiration in my early days of figuring this stuff out and we've compared notes on several occasions. Same goes for fluid and I, and many many more users of these tools.
These tools come in handy but require work to get a meaningful result. It would take me a long time to go over it and explain all they do, but I won't in this thread. I do have a thread which is almost 400 pages long that has a lot of this stuff covered, open for anyone to read and you'd be surprised how many have already done that.
There's a thread on the full range side of this forum that has helped a number of people get started with DRC-FIR. But it's still just a tool and any tool is as good or as effective as the user that determines what to do with it. It does tend to take time though, as we tried to explain earlier.
(*) you can even limit the phase EQ to a certain bandpass, for instance to only use it on that low end.
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It really follow our ability for localization: below 1khz our brain analyse ITD (phase delay) to determine localization, above 1,5khz IID/ILD ( Interaural Intensity Difference/Interaural Level Difference) and group delay are used.
https://en.m.wikipedia.org/wiki/Sound_localization
So yes your comment makes total sense to me: having the ability to modify phase behavior below 1khz to something 'linear' until 'natural' cutoff of system ( minimum phase behavior) is a powerful tool...
It took me years to understand why an approach like C.I.D. acoustic treatments ( Controled Image Design) was working as limited by design to 1khz and up.
Robert Walker didn't really explained it clearly in his BBC's white paper even if it's clear the redirecting of Early Reflection outside the main listening spot by reflection help our brain in interaural intensity difference treatment...
I would really like to apply phase 'linearisation' to a loudspeaker into a room designed around this principle, results could be outstanding imho.
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