Accidentally posted before I was done typing.
I think most of us would do things different than each other. That's part of the art of speaker design.
I think most of us would do things different than each other. That's part of the art of speaker design.
This thread reminds me of what not to not post here (DSP dial-in data)...
However, based on John Dunlavy's last interview at Stereophile and his admission that he was intending to build a fully active (DSP) version of his "Magnus" design, so it does hold some interest:
![Magnus2b[1].jpg](https://www.diyaudio.com/community/data/attachments/1366/1366373-0a6ebcac1ac67ba4dd14ff80f503cbb6.jpg?hash=mQQbfFqpti "Magnus2b[1].jpg")
Chris
However, based on John Dunlavy's last interview at Stereophile and his admission that he was intending to build a fully active (DSP) version of his "Magnus" design, so it does hold some interest:
Chris
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Interesting paper - but only ONCE - is music mentioned in the test used - and I quote:
"The conclusions of various tests have been that large enough group delay errors may produce audible errors, but when listened to music or some other real sound material, the differences are often inaudible"
I'm not saying that we should not make further effort to improve everything possible - but we should also consider the challenges every time we bring a new tool to the workshop. An active system brings many good things to the table - many advantages. But again, these are only advantages when used cleverly.
In most cases, people still battle with some of the most basic in loudspeaker design like resonances, break-up modes, filter slopes, power response, EQ, baffle design, room acoustics, over-all setup etc.
I could agree, that linear phase could be the icing on the cake, when everything else is absolutely done almost perfectly..... but until then... back to the drawing board
😉I do think that he may need some help on measurement technique. ( That's just an assumption based on what's been posted. ) What you suggested is good. I'm still asking questions trying to see what he's actually done so far. He may not want my advice either. I assume that he has made on axis measurements, but he has not actually said so. There are some fine points in making a good x-over that the OP may not be aware of. Or, since he has been at this a while, he may just not have mentioned some of the details, or maybe he has figured out other ways to get the job done. I'm trying to help if needed, but not force my design ideas on him. I don't think you are either.
In general I agree, the problems you listed have a much greater impact on the final result than the linear phase. The linear phase cannot eliminate the influence of the room on the sound, etc., etc. Therefore, yes, the linear phase is the icing on the cake, not the cake itself.
It is not always clear that some advice comes from practical experience. When you focus too closely on one thing, other things can suffer... (and not for want of trying, sometimes the practical compromises force it to be that way.)
To summarize the discussion, seems to me like the OP
- uses DF75 preset crossovers, so only cutoff freq and slope adjust, no level and eq https://www.accuphase.com/model/df-75.html
- uses no equalization for each pair of drivers/way to adjust baffle diffraction and cone resonances etc
- uses smaart to adjust delay for each way and perhaps spl response at listening spot (5 meters away!)
- we have not seen full system spl and phase response measurement...
- measurements at listening spot are corrupted with room reflections and modes
- to handle baffle diffraction and driver nonlinearities in acoustic domain (=equalization, requires quasi-anechoic measurements) like the original passive speaker did
- to check/adjust delays so that the work with the xo slopes (IIR)
- - measuring individual ways and matching delays does not take into account crossover-induced phase shift as part of a system
- the OP is measuring the speaker with set xos and trying to force step response to single peak by adusting delays radically.
- the OP is ignoring spl response and directivity
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Just use 4-5 measurements in the listening position, push a button and let DIRAC figure it out - voilà - you now have a perfect speaker. I'm kidding, of course 😉
Indeed. For laughs, here's what I would do (or have done just not so BIG). Three matched integrated amps (e.g. my TPA3221 bpl) plus a subwoofer plate amp with continuous phase knob for the bottom. With each driver-pair in LX position wired opposite polarity determine its combined null i.e. acoustic center offset. Mount the drivers using shims or custom adapters to ensure time alignment (as Dunlavy intended). Cover baffle surfaces with textured wool felt; add rounded moulding to cabinet hard edges. LCR notch filter to rid all drivers of resonance peaks within two octaves of intended range. Balance them with individual volume controls. And finally --
Set upper active crossovers all to 1st-order. By trial-and-error tweak each XO frequency up/down until phase is matched. This can be determined by listening to test music, and found/verified playing test tone at said XO frequency -- shifting driver offset would decrease combined volume, or undo null if reverse polarity. Done.
Other than present-day finer aspects of in-room response/directivity control, any technical reasons this wouldn't work in principle, everyone? Thanks.
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