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Improving Transition Range FR Swings in a 2.2 System

klettermann

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I’ve been doing a slightly unconventional room treatment experiment that I thought might be of interest. The starting point is an already optimized 2.2 system: Magnepan mains plus a time-aligned/EQ’d mono dual-sub array. Overall response is good, it sounds great in fact, but there have always been a few annoying irregularities through the transition/lower-midrange region that simply don't respond to conventional PEQ. No surprises there, I guess that's pretty normal and something most of us suffer from to varying degrees.

So, rather than asking the existing sources to do something they apparently can’t do, what happens if I introduce another independently located acoustic source with a different room response that can be used to compensate?

The experiment

I added a modest “helper” speaker (Magnepan SMGa) on its own DSP channel. It was placed horziontally centered against the front wall behind a large TV and angled up at about 45deg. Why like this? Cause that's the speaker I had and that was about the only place it could go. The aim was as a very limited correction source whose only job is to contribute where 2.2 system still has deficiencies. The existing 2.2 system remains the baseline. The idea is simply existing acoustic response + separately controllable spatial source to improve the region that more EQ can't improve. The "helper" had independent control of:
  • placement
  • bandwidth
  • gain
  • polarity
  • delay
  • PEQ
Below is the starting point. Blue is helper alone, just XOing to limit response to region shown. Red is mains response corrected to degree possible in same range while black is mains+helper.

1788227906523.png


Effect of Helper Timing

The helper is about 40 inches behind the mains so time misalignment would be expected. Rather than calculate I ran a series of delays in case of possible latency issues with the DSP stuff. See below.


1788228756261.png


There obviously isn’t one delay that phase-aligns two physically separate sources across several octaves. That isn’t what I’m trying to achieve. The task is to find the delay giving the most useful compromise across the particular deficiencies I want the helper to address. The broad optimum ended up around 3.25 ms in this setup.

Gain - Less is More

Below shows progressively reduced helper gain. The inital setting was obviously way too high. Less is more. 1/3 smoothing gives a better sense of the effect than 1/6.



1788229387931.png


Provisional Result after EQing

Below is the end result after a little helper EQing (almost). Black is the optimized 2.2 system alone. Red includes time alighned and level matched helper. Finally, blue is same as red but mildly EQing the helper. The wild transition swings are greatly improved. :D

1788267658044.png


So, the idea seems to have worked. Transition nulls that are uncorrectable by EQ don't mean the acoustic response itself is immutable. Adding another spatial degree of freedom can change the problem. This obviously has conceptual relatives in multisub optimization, Sound Field Management, Dirac ART/support speakers, etc. I’m certainly not claiming to have invented multiple-source acoustic control. What struck me as less commonly explored is extending the concept in a 2.2 system upward into the awkward ~100–400 Hz transition/lower-midrange region with a dedicated low-level helper channel.

There are also obvious qualifications. I still need to:
  • test several nearby microphone positions to see how much of the improvement survives spatially;
  • do substantially more listening with helper on/off;
  • examine the time-domain tradeoffs more carefully.
  • figure out other tradeoffs, such as stereo separation
So I’m not presenting this as “problem solved” or some astounding, breakthrough improvement. It’s just an experiment that turned out considerably better than I expected. The biggest question now is whether the improvement proves itself in spatial and listening tests. Comments, criticisms and suggestions welcome. Cheers,
 
Pretty interesting result graphs-wise but I'd expect you to notice some changes to the stereo image pretty quickly in listening... Or is it attenuated enough not to call attention to itself? I'd also love to see how this works if you try ART or Perfect Soup.
 
Pretty interesting result graphs-wise but I'd expect you to notice some changes to the stereo image pretty quickly in listening... Or is it attenuated enough not to call attention to itself? I'd also love to see how this works if you try ART or Perfect Soup.
Right, imagining and separation are my concerns too. I haven't been able to listen at all yet cause of travel but I'm eager to get back to it. I don't have ART or Perfect Soup. Maybe time to look at them....
 
Right, imagining and separation are my concerns too. I haven't been able to listen at all yet cause of travel but I'm eager to get back to it. I don't have ART or Perfect Soup. Maybe time to look at them....
They both have a free trial, I have vague ambitions of trying them out on my 2.2 system. I think they really only work to the extent you have multiple speakers playing under 150hz or so, but that applies here...
 
They both have a free trial, I have vague ambitions of trying them out on my 2.2 system. I think they really only work to the extent you have multiple speakers playing under 150hz or so, but that applies here...
Honestly, I gotta give it a serious listen before doing more. Setting up to do all that was a pretty major project - new MiniDSP Flex, USB interface and digital router. I'm also quite sure it isn't optimized as well as it could be. But - and it's a big but - I have no idea how audible all the transition zone chaos is in the first place, and I don't know if anybody actually does. So that's my next move.
 
Honestly, I gotta give it a serious listen before doing more. Setting up to do all that was a pretty major project - new MiniDSP Flex, USB interface and digital router. I'm also quite sure it isn't optimized as well as it could be. But - and it's a big but - I have no idea how audible all the transition zone chaos is in the first place, and I don't know if anybody actually does. So that's my next move.
Based on years of tweaking EQ I can say that ~5dB swings in that range are quite audible but don't jump out at you as obviously wrong... until you listen to a corrected response for a while, then put the deviations back. :)

But if you fast switch any of the filters in that correction you'll be able to hear it pretty easily I think.
 
Right, imagining and separation are my concerns too. I haven't been able to listen at all yet cause of travel but I'm eager to get back to it. I don't have ART or Perfect Soup. Maybe time to look at them....
I have a 2.2 system (co-located subs) and 2 additional subs I can include or not (2.4) and have messed around with manual REW/ Rephase, automatic REW, DIRAC live, DIRAC DLBC, and now DIRAC ART. For good sub integration and smooth frequency response and nice sounding bass it is going to be hard to beat ART. On the downside my manual DSP 2.2 system has a more defined "phantom center" compared to ART and the sub integration and EQ still sound excellent even if not as good as ART. I can switch back and forth easily and am torn between which I prefer but I have found myself listening more to the better imaging manual EQ recently. There is very little information about ART on dedicated stereo systems as most people using ART have AVR's and a MC system even if playing stereo content, in that case the phantom center is not an issue.
 
Very cool experiment, I've actually considered something similar but would have to free up a channel(s) on my 8ch DAC to properly implement. In the near future I'd like to implement a passive XO between mids and tweets to accomplish this - use one channel for another sub and the other for this maybe. I'd have to find/build the extra speaker though.

I'll confess that I now mostly use Dirac Art with my 2.2 system instead of my meticulously crafted filters. Subjectively they both sound equally good though my filters get extension below 20Hz but measurements clearly show Dirac handling sub bass and decay better.

Please do keep us updated with your listening. Highly recommend Hang Loose Convolver for instantly comparing level matched filters, there's a free trial.
 
Very nice work, but how the hell did you start with a 10dB dip round 100Hz with a 2.2 setup in the first place? That must have sounded very thin. SBIR from the Maggie positioning? Why not run the subs to 200hz?
 
Very nice work, but how the hell did you start with a 10dB dip round 100Hz with a 2.2 setup in the first place? That must have sounded very thin. SBIR from the Maggie positioning? Why not run the subs to 200hz?
Thanks. As for 100Hz dip, can you point it out? That certainly should have got my attention. System isn't thin at all, to the contrary. Maybe I goofed someplace in the scaling, smoothing etc? This was a lot of measuring, comparing etc, easy to lose track. But I think the overall point remains. And I'll certainly be doing a final recheck of it all, along with listening. Now I'm wondering if stereo helpers might be the way to go, but that's for another day. :cool:
 
Thanks to all for the comments. It was not only a fun project but confirmed to my partner that I've completely lost my mind. I do find a couple aspects of the project rather striking. The first is that I'm not aware of much (or any?) similar stuff being done in the 2-channel realm. In the multichannel/AV world, sure, but again not so much directed towards improving front channel response. But honestly, I haven't looked very hard, so I could certainly be wrong. The other is implementation - the many ways there seem to be to do the same thing, more or less. That whole topic merits more explanation.

The implementation issue..... There's ART, Hang Loose Convolver, HQPlayer, Audiolense, Acourate, Trinnov and probably more similar stuff. My problem is that my system is hardware-based. Routing, crossovers, delays, EQ and presets all reside in hardware, without requiring a computer or particular playback application. The alternatives all require either a computer-based processing chain or moving substantially into some proprietary DSP ecosystem. Neither is a simple addition to what I've got. No doubt the alternatives can produce excellent results, but the implementation seems greater than the remaining problems to be solved. So I appreciate the suggestions, but I think I’ll continue refining the system I already have.

Regardless, I'll follow up on all this and real-world listening. And FWIW, I can instantly switch the helper in/out in real time. :) More to follow.... Cheers,
 
What hardware do you use for the EQ, XO and delays?

Somewhere I've an old paper saved about using a 3rd speaker with 2ch but the sum of L/R is canceled or maybe the opposite. If I remember correctly it was supposed to increase ambience but no mention of EQ for that channel. I'll try to dig it up later.
 
What hardware do you use for the EQ, XO and delays?

Somewhere I've an old paper saved about using a 3rd speaker with 2ch but the sum of L/R is canceled or maybe the opposite. If I remember correctly it was supposed to increase ambience but no mention of EQ for that channel. I'll try to dig it up later.
MiniDSP SHD Studio for the mains and subs (stereo in, 4 ch out). MiniDSP Flex for the helper channel. The Flex still has 3 channels left for future fun. Nothing more esoteric than that. My ambience/presence/imaging/sound-stage/etc was already quite amazing beforehand, so no need for improvements there. This was strictly about improving FR through the transition range. Cheers,
 
@klettermann maybe 10dB was an exaggeration, but the black trace (before the helpers added), has a hole between 100Hz and 150Hz that may be 8dB or 6dB depending on how you draw your preferred response line. And another dip at ~220Hz. Either way, it looks like subs not running high enough, or mains not running low enough, or both running but out of phase. Or SBIR - if you've placed your subs near your mains they'll have the same SBIR issues.

Screenshot 2026-09-03 at 11.30.51.png

I'll stick my neck out and guess based on the "before" chart that shows duplicated dips at ~140 and ~220Hz that the main speakers are ~1.5m from the front wall.

Screenshot 2026-09-03 at 11.37.59.png

Hard to know without full details, but I would place the subs hard against the front wall (no SBIR), and run them to ~250Hz. I say this as an owner of dipole subs and speakers :-)
 
Last edited:
@mcdn Comments appreciated! I'll take a few steps back to try to flesh this whole thing out a little more. The helper story is just the latest in a series of ongoing tweaking projects. It might be helpful to review the larger path. DISCLAIMER: sweeps and diagrams were from various earlier work on all this, so consider them illustrative of the point, not definitive - justtoo many scans to pore through and I'm not organized enough. So here's the bigger picture.
-----------------------------------------------------
1. SUBS & PLACEMENT

Anyway, the first phase of all this was the dual-sub setup. See below, which also shows the room. The dimensions and placements are exact and the result of necessity (mostly doors). REW predicted it pretty well. Note massive null at about 58Hz.

No Sub Timing
image.png


Front Sub Delayed
image.png


This was borne out by the measurements, see below.

Sub Time Alignment
image.png


Blue is both subs, no timing, no EQ, no XO. Note horrible null (a few Hz away from Room Sim prediction, but the same thing). Orange is just time aligned at MLP. Null gone. Green is an EQ'd version of orange. I can't recall exact EQ strategy, but for sure it didn't go above ~120Hz cause I figured it would be negated by the XO. Regardless, all the transition garbage is quite visible.
And prior to all this I did a hugely extreme speaker/MLP positioning exercise. The room has a few annoying features - large width-wise soffit, a column and big TV/gear rack centered on the front wall. Speaker positioning did find the best positioning but several annoying problems remain and can't be entirely fixed without causing other, worse problems. So, at this point, the positioning is what it is.

2. MAKING EQ PRESETS

I won't elaborate much on this except to say that this is what preceded the "helper" project. The idea was to create a small family of preset EQ "voicing" sets along the lines of the silly matrix below. The scans show one of the versions of those.
1788443892362.png
1788444037686.png


A common feature of all those (and others as well) is the troublesome region in orange. The notches at ~230Hz and 380 are really baked in. No feasible positioning or EQing can budge them. That led to the idea of the helper - try to improve them, and the transition region in general, using an independent speaker not subject to the same positioning constraints as the mains. The 320Hz really seems high enough that fixing it would interfere with imaging.soundstage/etc, so I didn't consider it.

So that's the backstory to all this. And it's FAR from over. I view the results thus far really as only proof of concept. There's certainly a lot more to try without buying anything else:

  • Try to optimize helper EQ and system XOing and see how far it can go
  • Try the concept using small stereo helpers. The helper is a wide, diffuse, mono source. I've got couple of KEF LS 50s that might be able to do better in stereo but not damage imaging. ???
I'll carry on and post results as they roll in. Hope this helps clarify. Cheers,
 
Has anyone experimented with separately positioned, independently processed “helper” speakers to address persistent upper-bass/lower-midrange dips? My mains are Magnepan MGIIIas, with two separately placed subs already integrated using REW and miniDSP. The graph below shows the individual main-channel responses: red = left, blue = right, 1/6-octave smoothing. These are with subs off and no XO, so the resp9nses below ~80Hz can be ignored.

1789053662468.png


The channels have distinctly different trouble spots: roughly 230 and 384 Hz on the left, and 141 Hz on the right. These features persist through various EQ settings and further EQ hasn’t satisfactorily filled them. So then, the idea is to introduce another acoustic path using a helper, band-limited and shaped to contribute where needed. Each main would continue operating normally, with the helper supplementing its output selectively.

I already have a small Magnepan panel centered against the front and behind a TV. It has its own DSP and amplifier, ready for initial experiments. I’ve done some preliminary messing around, but my methods were flawed, so I’m starting fresh. I don’t yet have a properly tested result to report. One approach I’ll explore is using an inverted slice of the main response as a starting target for shaping the helper’s measured response, then adjusting bandwidth, level, delay, polarity and EQ against the actual acoustic sum. The magnitude inverse alone doesn’t account for phase, so its usefulness remains to be seen.

If this whole idea shows any merit I’ll try a pair of KEF LS50s as independent L/R helpers, each receiving its respective channel’s signal and addressing that channel’s deficits. I’ll check the individual pairs, the complete system with subs, and nearby listening positions, along with listening for changes in tonal balance and imaging.

So - Has anyone actually tried this at all? I’d welcome practical experience, relevant references, and thoughts about the method or likely pitfalls. I’ll post results as I go. To be clear. This is mostly for my own entertainment, using equipment I already have. But a worthwhile audible improvement would be a welcome bonus. Maybe more will be more. Comments welcome. Cbeers,
 
why make a second thread on the same topic?
 
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So this is the same concept as placing subs around the room to fill nulls, but at higher frequency.

The reason it works with subs (and even this is somewhat controversial) is bass is hard to localise in-room. I've informally found that I can kinda hear direction down to 65hz or so. But in general it's not that big a trade-off to have bass sources at "wrong" locations.

In contrast, doing this at 200hz or anywhere near it will be IMO really distracting and will create a strange stereo image.

I think this is one of those situations where acoustic treatment is probably a much better answer.
 
why make a second thread on the same topic?
Oops! It took me so long to get properly set up that I forgot all about it! I'll go back to that one. Thanks and cheers,
 
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