eTrap working

Last year I moved the BagEnd eTrap from the living room to a hallway corner near the second bedroom.  In fact, I had originally purchased the eTrap for that corner, thinking a passive absorber would be unwieldly there, whereas I could use a passive absorber in the living room.  One thing was I wanted to reduce the incredible bass transmission into the second bedroom.  I've been trying to make the second bedroom liveable, and when I first got subwoofers, the bass level in the second bedroom might well be 20dB higher than at the listening position because of household resonances and couplings.  And with the hallway wall rattling also.  I then spent about $3000 rebuilding the hallway wall (it's now 5 layers thick, with the outer three layers "suspended") and $2500 on a sound resistant door.   I didn't realize this direct approach was going to cost so much, I had previously planned just to buy the $1599 eTrap.  Finding that even all the above didn't help, I bought an eTrap anyway, just for this corner, but I had originally deployed it to deal with issues in the living room itself which now seem lesser anyway, perhaps because of better EQ adjustments.

A few months ago I had just moved the eTrap to the corner where I had intended to put it all along, but didn't have time to adjust until I got home from vacation in December.  I've tried various things since then (trying to deal with a 100 Hz resonance for awhile, but that proved problematic).

But mainly I've had one (of two) electronic traps set basically as low as possible, with as high feedback as possible, on the grounds that it's hard to absorb low stuff, so the lower the better.

But I was noticing tonight that in the second bedroom, a decent amount of 63 Hz was getting through when I was playing Briza.  I kept playing that track and testing different settings.  Just a tiny bit higher seemed not only to suppress the 63 Hz much better, but make it almost impossible to follow the bass line in the second bedroom, whereas before it had been quite easy to follow the bass line even with the door closed.

Strangely I had only moved the control up just past the "34 Hz" setting.  Previously it had been turned much below that setting, as far down it would go (not marked).

It was also easy to see how much attenuation was occurring right in front of the eTrap using my RTA, and that it seemed to extend to 63 Hz with the current adjustment, whereas 63 Hz was not being attenuated with the previous adjustment.

This seems to suit the purpose intended now, though perhaps more precise optimization would be better, but it's far better at suppressing bass in the second bedroom than it was before, because that leakage bass is mostly in the 39-70 Hz range.

Also showing that the thing actually works.

PB13 Ultra Port Tuning

When I restarted the left sub last month, I wasn't sure whether to set the Port Tune to 16 Hz or 20 Hz. So, to be conservative, I chose 20 Hz.  I believed I had one port open.

After reading the manual online, I realized there are only three approved options:

1) Sealed (choose "Sealed" tuning)
2) Two ports open (choose 16 Hz)
3) All ports open (choose 20 Hz)

One port only causes chuffing, which I thought I heard a few days ago when sweeping the oscillator down low.

So, I took off the metal grill today, and discovered I DID have two ports open.  But I had chosen the now incorrect 20 Hz tuning, so I reset to that to the 16 Hz tuning.

That resulted in essentially flat response from 30 Hz to 20 Hz on my RTA, instead of a slight rolloff at  20 Hz when I was using the 20 Hz tuning.  I saw no other differences.

The speaker works flawlessly this way, but the room shakes badly if the level is loud enough.  The 20 Hz tuning might not be a bad idea anyway, and I might have chosen it before for that reason (to suppress room resonances around 20 Hz) but for now I'm going to try the flatter response, especially as I have now leveled the bass to the midrange level.

There seemed little difference in the stereo correlated response.

Bass candy recordings like Grouse and Bass Mechanic sounded better than ever.

LR4 in-polarity

I was obsessing over this last week.  It seemed to be then, and up until a few minutes ago, that while the Linkwitz-Riley LR4 crossover has both drivers in phase at the crossover frequency, because each has 180 degrees shift from opposite directions, as you move away from the crossover frequency, the phase would unwind in different directions, so they would not always be in-phase at every frequency.

This is wrong, I finally figured tonight.  First of all, Linkwitz says specifically that both drivers are always in phase for LR4.  At every frequency.  (And I should believe him, of course.)

Second of all, it is true, the driver phase either wind up more (as you attenuate) or unwind back to zero (as you increase to max).  However, when you are moving UP from the crossover frequency, this means the HP is "unwinding" and the LP is "more-winding", and the HP winds toward leading, and the LP winds toward lagging, so when one is winding and the other is unwinding they are actually moving in the same direction because they wind in different directions.

I used to think the polarity reversal on one side of an LR2 was the magic that made the in-phase-at-every-frequency work.  Because of that polarity reversal, I reasoned wrongly, the phases move in the same direction as you move away from the crossover point in either direction.  But that's just a compensation for the crossover frequency itself, and it doesn't change the way the phases on each driver change as you move away from the crossover point.

Along with all this, I believed there were "in-phase" connections of LR4 and "out-of-phase" connection of LR4 that would work even better because less group delay.  I read that somewhere about some crossover, but it was clearly not about LR4.  I believe another JAES author (Bullock?) wrote about a series of all pass crossovers.  Linkwitz himself stumbled upon another one (the Dueland) which might actually be better than LR4 because more gradual phase changes.

Somewhere I think I read about delaying the LR4 Highpass by "360 degrees" presumably at the crossover frequency, to account for the fact the leading vs lagging of the two drivers.  Now I can't seem to find that suggestion.  I didn't try that much seriously before (I tried going in that direction a bit, and results were weird enough to make me think about doing a more serious investigation).

Well if that works for LR4, what about doing LR2 using 180 degrees of delay, rather than an actual reversed polarity?

Another Day, Too Many More Adjustments, Maybe

As I'm starting to write about the last measurement/adjustement/measurement/etc extravaganza I found myself locked into last week, I'm actually enjoying William Orbit, Strange Cargo II, largely helped by the dramatic increase in bass "dynamic range" because of the most recent set of adjustments, and other ones recently.

More and more I'm getting away from the "Room Curve" fudge, although I still leave some chocolate in places.

Exaggerated low frequencies are actually very problematical, for everyone!  The problem is, almost inevitably, regardless of much fretting to the contrary, the limit in how loud you can play is going to be in the bass.  That's where it gets distorted the fastest as you start cranking it up.  And with exaggerated bass, you can't even listen to the rest of the signal at a normal level, you have to turn everything down to accommodate the loud bass.  I've noticed that even recently on my bass torture test recordings like Bass Erotica.

Now, I did once use Rogers LS 3/5A, with a T27 tweeter, which can be blown, without too much effort.  That's when I went off the ranch and got a pair (and, in my dreams two dozen) Dynaudio D21AF tweeters, with 600W power handling capability and response to 40kHz.  I now use those as my living room supertweeters, along with the rest of the LS 3/5A box and unused Bextrene woofer, because building some other box will take a lot of time and effort.  I know they're good to 22kHz or so my phone shows me that, then apparently cuts off because of it's 48kHz sampling rate.  Those Dyanaudios haven't been made in decades and are hard to come by.  Newer Dynaudios, even the $1300 ones, aren't as flat at the super top end, I think mostly or entirely because they've scrapped the 21mm form and now make 28mm only.  Technical improvements may extend the 28mm tweeters to 40kHz but usually only 30kHz or so, and not as flat.

But in many if not most cases now, the practical limited of bassy recordings is in the bass, and hence the bass tuning, which I've recently lowered again (in a serious of adjustments, along with a slight rise in the right channel, which had itself been the subject of a forgotten experiment, as I think I may have written about a few installments ago).

Anyway, the new adjustments are sounding great.  I'm sure they could be better, I just had to stop somewhere.  I was losing track of how many different things I was doing, and thereby illustrating one of my own key weaknesses.  I may explore a lot of things, but in the end they may all get jumbled up, and with no rhyme or reason I may end up just cancelling the whole mess and starting over.  Or, nowadays, I can roll back to a previous set of settings, which is especially made more certain by taking pictures of them.  (I can save them too, but in my experience, I'd take a picture first, because going through the "saving" process and making one mistake can overwrite your settings, leaving  the latest discoveries lost to mind and history.)

I first started with strong wish to verify the goodness of the notch in the midrange at 119 Hz.  This seemed to curious raise the response in the right channel, by removing some cancellation effect I suppose.  On the left channel, it seemed to have no effect.  I needed to verify both of these claims to feel happy with a 119 Hz notch.

Sweeping was how I set the 119 Hz notch in the first place--that's how it sounds the smoothest and flattest (best of all at -15dB notch actually, but I waffled that down to -8dB originally).  In the right channel, listening with or without the subs.  Switching the notch in and out seemed to make no audible difference in the left channel, as if the panels were contributing nothing due to overdamping (however, I do not remember if I swept the left panels by themselves to test that).  This all fits my theory that the damping pad support is broken in the right speaker.  As a result, it generates a strong out-of-polarity resonance at 119 Hz from the damping pad vibrating backwards.  The right speaker may have a different problem, overdamping at 119 Hz.

And, perhaps even truer explanation, but my guess has been merely a coincidental effect, has been the room mode circa 119 Hz which causes maximum cancellation in the center of the room where the listening position is, and a slight augmentation at the backwall.

So I'm getting some loss at 119Hz at the listening position from multiple mechanisms, but both overall and in the right channel by itself, the 1/6 octave response is flatter with the 119 Hz notch.  (In the left channel by itself, it curiously makes no difference.)

I "proved" that, though what I've actually done is transfered the interpretation to the visual domain, rather than eliminate interpretation altogether.  Which is why I must show the graphs (even for myself--and I know I'm not trying to lie to myself--because I could have interpreted wrongly).

I've also proven that with sweeping, though I can't show that, it's just a memory, already becoming more uncertainly.  I believe I veried the "both channels at once" part (because I now always use the Y adapter with the oscillator so it drives both channels), I just switch the channels by changing their relative levels in the Tact "Level" menu.  I set the unwanted channel to -99.0dB.  But I'm not 100% sure, much like sometimes when I take my bedtime medicine without the daily pill box.

So that's the advantage with things like FFT.  Measurement tools that work over time and capture the result to a permanent form.  That used to mean paper but now electronic is preferred.   They can be checked and rechecked ad infinitum, with no disadvantage  in the future from when it happened, except that when it was happening you could have run a different measurement in some way, either to fix something or verify something else.

But that should not be confused with intrinsic accuracy, the radical accuracy of getting to the root of the resonant phenomenon, and cancelling it out with exactly the right Q and frequency response.

This is NOT AT ALL like adjusting the knobs in a graphic EQ to get flat response.  First of all it never works, and second of all it always makes some things worse (I conjecture, however I have only limited experience with such things myself).  What is bad is well corrected, and what was good is now awful.

In principle, the oscillator method and parametric EQ method I use is no different from what a program like REW does.  It does that thing of matching IIR filters to the most important anomalies in sequence, for as many parametric EQ's as your system supports.  What could be wrong with that?

What IS wrong with that, and in principle must be wrong with a system like that, is that a finite signal must be used.  With an oscillator, a virtually infinite signal is used.  This gets to the root of the resonance.

Well not so, one might argue, one isn't actually sweeping THAT slowly.  One may (and often does) hold a particular frequency for many seconds if not minutes...that is still not infinite.

Well many seconds at some particular frequency is Far and Away different from what any FFT system (that I am aware of) does.  The typical chirps, even 10 second scans, are nothing compared to holding one particular frequency for seconds.

And that's why system that can't seek can't get to the root of things.  It's not to say, exactly, that this couldn't be automated.  But FFT does not particularly help in some ways...it's by definition a finite system...

The human with an oscillator can sweep around looking for problems.  Problems are not usually hard to find when sweeping.  Usually, the first sweep shows how incredibly horrible things are, it's then just a matter of deciding which thing is worse (that is, of course, a subjective decision) and then correcting it with a parametric filter that cancels the problem a much as possible, and then deciding which is the next worse thing (another subjective decision).

The correcting part may involve VERY SLOW sweeping, and often sustained tones where deep resonances are found.  This is only feasible if someone or some thing has made the decision about what the anomalies are and which is the most important one FIRST.  THEN, it can be done, but with virtually infinite stimuli.

But still, you must rely on memory, of which I have little, and judgement, which is always in doubt, and it's nice to have pictures too.

Here's the picture of the correlated pink noise response without the notch.  Notice there is a severe depression around 120 Hz.  That depression is significantly larger than the depression around 55 Hz.  

Correlated response, No notch  added

With EQ, there is usually nothing I can do about depressions caused by model suckouts.  Those are only responsive to radical changes in listening position (say, from the middle of the room to the back wall) or sometimes other room alterations...dampening and absorbers to reduce standing waves at the modal frequency can sometimes curiously raise center-room response.  (Absorbing sound to prevent reflective cancellation which causes reduced sound elsewhere.)  Some of the depression around 110 Hz (actually, around 119 Hz) is caused by a modal suckout, which somehow applies especially strongly to the panels, less to the subs, the reverse of the usual thing.

Now curiously, while the response at the listening position is sucked out at 119 Hz, the sound between the panels and the subs is incredibly loud.  This is partly due to room position, but I began to suspect something different, and discovered that turning the panels off caused the response at 119 Hz to rise!

So, what if I could just cancel out the 119 Hz panel output in such a way as to reduce whatever is causing the panel operation to cancel the subs?  Of course I can with a notch filter, and I tuned a notch filter to flatten the output around 119 Hz when sweeping with an oscillator.  This was almost, but not quite, as narrow as I could make it.  (I first described this discovery in the last post, but I didn't have pictures yet):

Correlated response, -8dB notch added at 119 Hz

The depression around 110 Hz is visibly flattened, so now it is significantly less than the depression around 55 Hz.  I could have also shown (or at least recorded, as I didn't bother, but proves that non-graphic methods are possible if you record a boatload of data, as I never do) how the SPL response in the listening chair rose as the notch was increased to the full -15dB I have available.  However, the above is with a "compromise" conservative notch value of -8dB.  Without any notch, the response of the phone/spl meter in the listening position was at background level 24dB lower without any notch!  I felt this brought response to within 3dB of what was available at -15dB, without going all the way and suffering side effects that the steeper notch might have.  But if I had been optimizing the measurements, including the graph, I would have gone for the full -15dB notch.

Now, one could argue, this was just a random chance, that the notch performance looked different.  So I did a second measurement, which has the same visible characteristics regarding the 119 Hz depression.  It is less than without the notch.

Correlated response with -8dB notch, repeated

This visibly proving the curious effect that a notch (in the panel output only) is increasing output.

Now it was clear to me that the response in the notched channel itself was much better with the notch, so I didn't graph that.  But what about the response in the alternate channel.  Would that now get much worse with a notch it didn't need?

The answer was, as I reported in the last post, the response with and without the notch in the other channel was identical.  In either case, the response is sucked out around 110 Hz.  The notch appears to change the already suckout at "110 Hz" (actually 119 Hz) in this channel by a mere 0.1dB.  As if all the significant output of the left channel at 110 Hz is coming from the sub anyway, so the notch essentially has no effect here, while it is extremely beneficial in the other channel.

Left Channel, with -8dB Notch

Left Channel, No added 119 Hz notch

Sweeping didn't reveal any finer detail than this, both just seemed to have an identical suckout around 119 Hz.

With the previous days labors now verified by graphical means, I proceeded to make some more changes, that also looked and swept good.  First of all, emboldened by the null effect of the -8dB notch at 119 Hz, I went ahead and increased it to a -12dB notch, which continued to flatten the right and combined responses with no effect on the right channel, it appeared.

To the left channel, I tried to suppres the circa 80 Hz peak better, as well and flatten the curve around 32 Hz.  I added an 80 Hz notch (or did I do that last time) and made tiny changes to the existing notches and then ultimately added a low Q 28 Hz notch which was not so much a tuning of the existing resonance as a tuning of the ultimate desired response, sounding nice all the way down to 16 Hz.  Otherwise, the left sub might need some port retuning, I need to look into that.

This is the resulting left channel response afte the latest adjustments:

Left Channel, after latest adjustments

Notice the short but slightly tilted window all the response falls within, including the bulge around 32 Hz, and the narrow depression around 110 Hz.

The stereo corelated response also looked a bit better, now with a similar short window of response.

Here are the latest adjustments which accomplish this visible and sonic magic

Subtracting to Add

I happened to come across an RCA Y adapter laying on top of the living room power conditioner as I was cleaning the right subwoofer and environs on Tuesday.  I was carefully vacuuming cobwebs and dust bunnies from between the woofer and the metal subwoofer grille, then I used a dry microfiber cloth to lightly brush off the remaining dust from the SVS woofer cone which appears to be covered in a felt-like substance (I think it might be a metal cone also).  I'm not sure if water could be used for this cleaning or not, or even if microfiber is all that safe, but I hoped it would be ok for a light dusting.  I was puzzling the 120 Hz "notch" which actually seemed to have a strong buzzing quality, even more like a strong modal resonance in the alcove between the back of the right Acoustat 2+2 and the right SVS PB13 Ultra.  I was also neatening the junk on top of the power conditioner in that area.  My cleaning did seem to help the buzzing, actually, but not eliminate it, and I think I'm going to have to remove the now heavily damped metal fireplace and rebuild the front wall to fix it, something I've been thinking about doing for years (but where to move the stereo in the meantime???  The organ and junk had to be removed from the Gym, to permit the Gym to be used as temporary holding space for the living room stereo.  The old organ and junk were finally removed last year, after 6 years of procrastination.)

I must have put that Y adapter there precisely for use with the oscillator, for stereo sweeping, which I had not done yet.  Ahah, one more kind of sweeping I can do!  I figured right away that as soon as I started stereo sweeping I'd identify issues that required mono sweeping again.  I was right.

Immediately I was back examining the 120 Hz suckout in the right channel.  It was quite noticeable in stereo (as in the stereo correlated pink noise response).  But then I also noticed that the bass was seeming quite light.  Had the subwoofers died?  Ooops, I had the subwoofer EQ turned to a different input, which is my tricky way of turning the subs off for testing the panels by themselves, which I must have been doing the day before.  But since then I'd spent a day or so listening to Marie-Claire Allain play the Complete Bach Organ Works, in background mostly, and marveling how good it sounded.  With no subwoofer and the panels rolled off at 100 Hz.  We audiophiles make these mistakes all the time, and in my case I'm all for pointing it out how unreliable our hearing is that major issues can go unnoticed for days (weeks, years, or even decades, I can show in my own amazing audiophile history), while we nit pick over small stuff that may be essentially impossible to hear.

With the subwoofers back on, it was now clear that the subs didn't suck out as much as the panels.  Something weird seems to be going on in and in back of the right panel at 119 Hz.  It's buzzy and loud, and apparently phase reversed from the subs.  If the subs are on without the panels, and then I turn the panels on, the level of 120 Hz at the listening position goes down to nothing, over 20dB down.  Maybe 30dB.  Now that's cancellation!

I could (and did try) to fix the problem by boosting the subs at 119 Hz.  But that didn't go so far, and made nodal resonances at other points in the room, such as in the doorway to the kitchen (where 119 Hz gets VERY loud) even louder, by the same amount as the boost (or seemingly more).

But what actually seemed MORE effective, was notching out the panels at 119 Hz.  And curiously the more I notched out 119 Hz in the panels the louder it became at the listening position, apparently because of reducing the cancellation of the more effective subwoofer drive.  It also seemed to reduce the buzzing, as if I was suppressing an anti-phase resonance in the panels (that's not my theory, exactly, but there may be some issue like that involved).

Dialing in the full -15dB available in a 1/10 octave notch at 119 Hz reduced the 30dB of cancellation to a mere 3dB of cancellation.  And it didn't seem noticeable when sweeping the speaker.  Weird stuff goes on at 119 Hz in any case, but with the -15dB notch it sounded only better than at lesser levels, not worse.

Still, I'm disinclined to apply such radical EQ.  I looked for the level where I'd get "most of the benefit" without having to go all the way.  And I found I could get within 3dB of the best available performance (at -15dB) with a mere -8dB, so I decided to go with that.

This is very strange.  I'm fixing a suckout at 119 Hz in the right channel by appling a notch of -8dB at 119 Hz to the right channel panels.  Strange but it seems to work and I get even better elimination of the notch with -15dB.

There is obviously something (or many somethings) very weird going on here, possibly including the acoustic crossover not matching Linkwitz-Riley LR4 alignment very well because of driver resonances AND other EQ corrections!  Also imperfect time alignment.

LR4 itself is a little weird.  It's not as "perfect" as LR2 perhaps, though it doesn't require polarity inversion on one side.  LR4 is strange because you can use it either with or without polarity inversion.  I think I like it better without the inversion, and it's one of the reasons I like LR4 (no polarity inversion required).  But Linkwitz liked it better WITH the polarity inversion, because it has less group delay that way.  Hmmn.  I have always wondered if no-inversion LR4 has regions where the drivers are not always in-polarity as they are with LR2 at ALL Frequencies.  At distant extremes, the drivers are in polarity, but near the crossover the relative phase twists around in opposite directions for each driver, further complicated by non-coincidence and non-uniformity-of-radiation.

These issues are worthy of serious investigation!  However, for now, I'm using the -8dB notch on the panels, which increases the near total suckout at 119 Hz at the listening position to a fluctuation.

Sadly, I do not believe I can add this -8dB notch merely to the right panels in my current configuration, which is stereo-linked for the panels.  I'm not sure I need to do stereo-linked anyway, but it has always seemed convenient for adjusting the panel issues, which aren't as position dependant.  There had not appeared to be a serious suckout in the left channel at 119 Hz that I remembered.  For now, I swept the left channel with the -8dB notch in place and surprisingly it sounded fine.  Perhaps the notch makes it better too.  Obviously I should re-check this also.

And I should remember I'm applying such a hack and re-check if it continues to help after future changes and adjustments.

I ought to try LR4 with polarity inversion.  I don't recall even trying it, I just didn't like the idea of polarity inversion.  But it seems to me now that only LR4 with polarity inversion yields the central benefit as LR2 which must have a polarity inverted...both sides remain in polarity at all times.  Without that critical inversion, the relative phases must be twisting around, and frankly right now I can't even fathom how it's even supposed to add up to flat response.

Perhaps much of the last 10 years of fiddling with EQ's around 100 Hz is largely to compensate for using a defective crossover alignment...

Using the Y adapter is nice, because I can switch channels using the remote (through the Tact Level Menu, reducing alternate channels to -99.9db) and compare channels.  It was clear the right channel was somewhat lacking around 32 Hz, as I remembered.  All the boost shown in the stereo response graphs was from the left channel response nearly exploding around 32 Hz and down to below 20 Hz, and not sounding all that good either.  I dialed in a -4dB 3/4 octave valley around 28 Hz to tamp this down.  It could be tamped down more.  This allowed me to reduce the notch at 40 Hz to -5dB.

More issues in this channel as well.  Is the subwoofer tuning EQ correct for the current one-port-open?  Is one-port-open a good idea in this channel?  I already went to sealed in the other channel, but I sorta liked the combination of the two in different channels.  I thought I liked the left channel often filling in extra whoomph otherwise unavailable from the sealed configuration.  Was I mistaken about the effect of that overall?

I'm noticing a lot more whoomp in recordings like Grouse We want to be loved.  How much this is due to the very last change, the 119 Hz notch which seems to boost overall response, I don't actually know, but it made me decide to increase that notch to -12dB (I know it continues to measure better, in the right channel at least, all the way out to -15dB, and that gives the flattest listening position response to 119 Hz, only about 3dB below baseline, which is what you get cancelling out one part of the signal--the panels--and leave the remainder).

In addition to a curious property of the in-polarity LR4 connection, another concern is the damping in the back of the Acoutstat panels, which sometimes becomes loose with age.  That could be the very element which begins to vibrate out-of-polarity at 119 Hz because of a resonance at precisely that frequency in the panels themselves.  Chalk this up as one prototype theory in which a panel notch at 120 Hz improves the response greatly at that frequency, at least in the right channel.

It may in fact be that resonance at 119 Hz and/or others is partly reponsible for the rattling/grating side I hear mostly in the back of the panel on certain pieces of music, notably a track from Grouse.  I can eliminate that rattling with a wide notch of 1 octave centered at 250 Hz, or better yet with -15dB sliders from 120Hz to 630Hz, possibly with some middle sliders back at zero.  The rattling is inconsistent and history dependent, making it hard to figure exactly.  But some reduction at 120 Hz always seems necessary, and I'm now already reducing the panel response by -12dB by default, which may reduct the rattling somewhat by default.

Sadly oscillator sweeping has not yet revealed a single frequency for the rattling, which sounds like the rattling of stator wires, possibly because an attachment has come loose.

Back to Square One

I couldn't let it go.  I got back to sweeping the oscillator, which now makes it clear that the 120 Hz depression is still there (despite my having eliminated the nearly useless 140 Hz notch in the subwoofer drive).  I may have made it better but it's still almost as noticeable when sweeping.

Curiously, the 120 Hz depression appears in both the subwoofer response AND the panel response, and it exists only or primarily at the center of the room where the listening position is.  So it's a modal depression that's stimulated by a tall dipole as well as a floor hugging monopole.  Dipoles manage to avoid or at least mitigate most room modes, but curiously not this one, it seems just as bad in either response.  I think, however, it might be coincidental that the dipole-backwall cancellation happens at the same frequency as a room mode.  So it's possible there are different causes for the 120 Hz depression in the dipole response from the subwoofer response.  But the fact that a depression happens in both means I can't boost either one to correct the other.  This problem is NOT correctible by EQ.  Fortunately it doesn't actually seem that large.  It's very localized around 120 Hz and unless I'm sweeping slowly I might not notice it at all (which is why, on the first night, I didn't notice it at times in either panel or sub responses, but actually it's in both).

The "1 kHz suckout" that appears in the right channel  RTA graph seems non-existant in oscillator sweeping.  Response seems audible flat from 250-2.5kHz, and on up through 15kHz where my hearing cuts off.  My system response (as measured by phone) seems to cut off about 22kHz.  That possibly shows the phone is sampling at 48kHz.

I also cancelled the first 100 Hz notch filter in the right subwoofer drive.  I appear to have had broad and narrow notches at 100 Hz, with the broad 1/2 octave notch intended to smooth the broad area around 100 Hz, but that makes the 120 Hz depression worse also.  I do in fact need a deep narrow notch just at 100 Hz but not much higher.

System response may now be about as decent as I can get it with sweeping and a handful of parametric EQ's.  It's probably time to move on to time alignment tests.

One exception is I might see what happens if I eliminate the 32 Hz bulge.  I didn't have that "problem" with the 1+1 speakers, even though in both cases the panels are crossed over at 100 Hz.  It may partly be because I've gotten more agressive with resonances in the 70-100 Hz region.  I used to excuse those as "room curve" but then it seemed like the bass got much cleaner if I eliminated the resonances underneath that curve.  In effect, I've whittled my way back to the bottom of the frequency response curve making it flat, where superficially the added boost remaining may seem helpful but may not be helpful with all music.

It sounded just fine playing the complete organ works of JS Bach played by Marie-Claire Allain today.

Working out the 120 Hz notch

I knew I had at least one more day of sweeping and adjusting ahead of me, as when I last left off I had discovered the apparent notch I believed to be at 110 Hz or so in the stereo-correlated pink noise response, was actually being caused by the right channel.  I knew that after I fixed the 90 Hz notch in the left channel, after which it was quite flat in bass, the subject of my last post.  So any remaining notch had to be in the right channel, even though it was not fully apparent in the right channel response itself.

I went through a series of false theories before zeroing in on the apparent cause, fixing it, and measuring the results.

My first theory was that the notch at 110 Hz was being caused by rear wall reflection and cancellation.  Such is my tendency to view results through the prism of my own theories, that my first sweeping 50-150 Hz on the subs seemed to verify this.  The sub seemed quite flat.  Actually I hadn't turned up the level enough to be very sensitive to changes.

So figured I should fix the problem by boosting the subs to fix the panel response, and I tried that.  Sweeping had shown the suckout to be at 120 Hz, somewhat above the crossover point, but still permitting at least 4dB of correction by boosting the subs with a high q resonance at that point.

Indeed, it seemed to help.  Only after much further investigation I decided to listen to the panel response without the subs (what I should have immediately done to fully test the backwave cancellation theory).  Only then I discovered there was no "hole" in the panel response around 120 Hz.  They were smoothly rolling off down to the -6dB at 100 Hz, and that was about all.

Redoing the bass sweep, I finally heard the suckout at 120 Hz.  Now I noticed that this frequency was quite modal, and the listening position was indeed reduced about 6dB from the backwall position.  I tried fixing this by tuning my eTrap to 120 Hz, which is technically out of it's range, but I tried and got a barely measureable reduction right in front of the eTrap.  But it made at most 0.1dB difference at the listening position, so I decided to abandon this correction.

It was sometime after this I noticed something.  I had previously dialed in a 140 Hz notch into the subwoofer response.  As this was above the crossover frequency, I felt quite cavalier in a huge reduction, possibly not very well tested.

So I tuned out this 140 Hz notch, and it turned out to be entirely unnecessary.  There was no large peak in the sub response around 140 Hz, so it was hardly deserving of a 12dB notch.

Removing this pre-existing 140 Hz notch, with a width of 1/3 octave, fixed the problem at 120 Hz and allowed for much flatter response in the right channel and the elimination of the apparent 110 Hz notch in the stereo-correlated response.

I made a new correction at 80 Hz which seemed to be useful and some other fine adjustments.  I held to the process of initial adjustment with oscillator sweeping, which seems to help avoid invalid adjustments.  Adding in new filters on the basis of pink noise measurement is asking for trouble.

Bottom line: the problem was caused by pre-existing currently (and/or previously) invalid adjustments which are now fixed.  The bass is now quite sonically flat in both channels, confirmed by measurements and playing Rebecca Pidgeon's Spanish Harlem.

Right Channel

I've never noticed this before...now there does appear to be a suckout at exactly 1kHz!  Worthy of more investigation.  I doubt any of today's changes contributed to that.

The stereo uncorrelated bass looks as flat as ever, with just a wiggle around 78Hz I tried to reduce a bit today.

That wiggle around 78Hz, with gradual dips on either side, is simply exaggerated a bit in the stereo-correlated response.  That fact and this graph looks excellent.

Here are the Right Channel EQ's that make this possible.  I didn't even remember the 60 Hz EQ, but when I removed it sure enough things got worse (and the dip around 55 didn't go away).

Another Day, Another Adjustment

Today I decided to tackle that dip in the living room left channel around 100 Hz.  I assumed that this was also related to the dip in the stereo-correlated pink noise at 111 Hz.  (This assumption now looks wrong.)

Crucial to my adjustment method is what I like to think of as "adjust-to-model."  In principle, I am determining faults in the system performance, and either fixing them (if possible) or "correcting" them through some kind of sensibly determined system adjustment.  The principle is fairly loose, I often don't know exactly what is causing this or that bulge or dip in the frequency response, I might simply label it "unknown resonance."

This is a vastly different philosophy, however, compared to the what-I-view-as-naive adjust-to-measurement school, which now seems to dominate the professional sound industry.  Endless devices over the past 40 years and still automatically measure your system.  Most take a fairly primitive adjust-to-measurement approach.  Actually, my TACT units are way more sophisticated than that, and I still don't trust them.  TACT uses a proprietary hybrid approach.  I'd prefer things be completely out in the open and customizeable.  Failing that, as all automated systems are, I'm fine with fully manual. I take measurements, devise some kind of model, and correct it.

In this case, I was convinced that the circa 100 Hz dip in the left channel was caused by an error in the delay compensation.  I spent several days over the past two years fine tuning the delay times for woofers, panels, and tweeters.  Most of this time was spent optimizing the right channel.  When it came to the left channel, I had little choice but to accept the same values, as until a few days ago I hadn't figured out how to set different delays in each channel of my Behringer DEQ's, which is the only place I can do it currently.

But now, I've figured out how to do that, so I got to work right away twisting the delay time for the left subwoofer only, to match it with the panels so there would be no "cancellation" I assumed was causing the dip at 100 Hz.

Well, no such luck, I turned the knob forwards and backwards by several milliseconds and there was no signficant change in the level as measured by my phone in the listening chair (I usually haven't been so exacting, I might even just listen to things on the floor and assume that's close enough.  But more and more it isn't.)

So if this notch near 100 Hz was NOT caused by cancellation, what was causing it?  It probably wouldn't be a room mode, or it would affect the other channel, and in the other channel I'm actually appling a large parametric EQ cut at 100 Hz (something I tried removing to make the stereo pink noise response better...but unflattening the response in one channel to make the stereo pink noise response better is not the path to Hi Fi...it's making the stereo balance at those frequencies Even Worse...as I decided quickly a few weeks ago after naively trying this adjust-to-measurement).  I immediately discovered the weakness was in the panel response, not the woofer response.

It didn't take too much thinking to begin wondering if it was boundary cancellation from the rear wall behind the Acoustats.  I though about how much distance the sound wave would have to travel at 100 Hz for 180 degrees of phase change.  That would be 1/2 wavelength, or about 5.5 feet.  My speakers are about 3.5 feet from the wall, making the sum about 7 feet.  That's pretty close, and perhaps all the extra stuff piled in this area reduces the "effective" distance.

I got out the oscillator and did some, well, lots of sweeping.  Note to self: despite my recent calibration, I can't really trust the markings on my Genrad oscillator.  I ultimately determined frequencies of interest by trying to cancel them with 1/10 octave notch filters on the EQ.  From this, it became apparent that the frequency of the most obvious notch is actually around 93 Hz.  That would correspond to about 6 feet.  Perhaps that's actually the effective "average" distance of membrane to wall given all the junk in the corner, or some other influence is bending the cancellation notch, but the measured notch seems too close to where the cancellation notch should be, and where the cancellation notch should be, perhaps around 78 Hz, there's actually a measured peak (around 81 Hz) that's I'm cancelling for (see later view of all left channel eq's).

In any case, given that delay has no effect, and the speaker positions are almost entirely determined by other considerations, including practical considerations, there's not much I can do except apply EQ to this notch anyway.  AND, since it's a notch in the panel response, it's essentially pointless to try correcting it in the panel EQ (and I only have mono EQ for the panels now, anyway).  Nevertheless, I thoughtlessly tried correcting the notch in the panel response, and it was nearly useless, with 6 dB of boost causing about 1dB gain in the measured response.

Fortunately, since this is very close to the 100 Hz crossover frequency (in fact, 93 Hz is technically IN the subwoofer range) I can, should, and must correct the problem simply by boosting the subs.  So, I simply added a high Q peak to the sub response at 93 Hz.  And it seemed to have considerable effect, though nothing like 1-to-1.  I didn't try to cancel out the notch as measured at the listening chair, but I raised the level a few dB.  Above all, one doesn't want to overcompensate, and especially when adding boosts!

Even with 1/10 octave bandwidth, setting this boost higher than a few dB seemed to make 100 Hz (where I previously thought the notch was) too loud.

I proceeded to spend 90 minutes sweeping the left channel 20-200 Hz, and tweaking existing notches mostly.  It was clear I needed to add a new EQ notch around 80 Hz, so I did, taking much of the burden off the notch at 71 Hz which I could then lighten up a bit.  Sweeping an oscillator by hand is FAR more internally revealing than even looking at a 1/6 octave RTA display.

In the end, I think I made the left channel bass sound pretty smooth.  I then looked at RTA and decided to make some additional refinements, then back to oscillator, and ultimately back to RTA.

I did the best smoothing job on both the swept and 1/6 octave measured response with 90 minutes of sweeping and measuring.  The result being this

There still appears to be a slight depression at 100 Hz.  But this slight depression is not obvious when sweeping the oscillator, in fact 100 Hz "sounds" louder than frequencies below it.  In my interpretation, we are looking at essentially flat response from 50Hz to 20kHz, within 2dB (when averaged by usual methods) except around 30Hz, where the additional weight may be helpful.  I've almost always had a far more elevated bass response than this, only getting enamored with "electrostatic bass" in the last couple years, in other words essentially flat "room curve" except now below 40 Hz.  Also notice an extremely gradual "tilt" downwards with frequency.  Many designers over the years have come to appreciate this downward tilt as being sonically beneficial.  In my case, the downward tilt is extremely small, about 2dB down at 10kHz (it used to be identical to 1kHz until I added a new EQ there last week, to make the highs flatter from 3kHz-20kHz).  As good as it gets!  (Well, for now anyway.  Until tomorrow's adjustment.)

You can see the plateau from 90 Hz on down which led me to add a second notch around 80 Hz as well as the existing one at 71 Hz.  There was a larger roller coaster without notching 80 Hz down.  When sweeping the oscillator, you hear the exact inflection points (even if you don't know exactly what they are with my not-remaining-calibrated oscillator) and can sweep the EQ's as narrow as possible to cancel peaks resulting from modes and resonances.

I am always disinclined to provide "boost" for various reasons.  Since I run my digital processors close to 0dB there is essentially no extra headroom.  If there was a pure tone at the exact frequency, the boost could cause digital clipping.  However, at 93 Hz, the subwoofer drive is already being attenuated by about 5dB because of the LR4 crossover at 100 Hz.  So I think adding back in 2dB of boost is fairly (if not perfectly) safe.  (Perfect safety may require as much as 6dB of headroom WHENEVER you apply a boost or do anything tricky, then you can be sure sample-overs or other peculiar waveforms aren't going to cause clipping either.  Countering one EQ against another isn't perfect protection from clipping, I've discovered on numerous occasions.  I'll have to see how this current boost goes in practice, but it seems like it ought to be mostly OK, with possible clipping once every month of Sundays a small price to pay for solid and true bass with no annoying notch near 100 Hz.)

Here are the latest left channel EQ's:

Sadly, all this work didn't seem to remove the dip or notch in the stereo-correlated pink noise response for both channels.  So, all this work, and I didn't address that problem at all, which seems to be above 100 Hz and not below it anyway.

Adjustments and Discoveries

That dip in the left channel frequency response around 100 Hz was bothering me.  It seemed this might result from improper adjustment of the time delay between woofer and panels.  I played pink noise on left channel only and seemed to improve the situation slightly with some adjustment.  Then I played stereo and the dip got worse.

At first, I believed this was an old problem I'd never solved before (but, I solved it tonight!).  It turned out to be another problem I would shortly discover.

The old problem was that the woofer to panel distances are not equal for right and left.  There has always been insufficient space to do that.  However, over time, the direction of the difference has changed.  Originally, the right panel and speaker were the closest.  But when I got the 2+2's, I had to squeeze in everything more away from the door, and not it's the reverse.  The right panel and speaker are now the closest.

But for many years, it has seemed that I could NOT set delay times differently for the left and right channels in the Behringer 2496 DEQ's I use for both crossover and EQ.  I typically set the delay time after experimentation in the right channel, and then just accepted whatever resulted for the left channel.  Sometimes I tried phase reversal of one unit or the other, and sometimes that phase reversal was required (or not) because of how I had most recently replaced a defective plate amplifier in the subwoofers.  There is a LONG history here going back 5 years and more if you count the time I was using DCX units which did allow me to set the delays differently.  Finding the polarity adjustments on the subwoofers to have weird results, I tended to use an XLR polarity reversal device.  I was still using that by mistake until at least November.  Currently I am not polarity reversing either subwoofer as I find the two subs sound out-of-phase when I do that when I play them with panels turned off, and I believe that to be an important requirement (even though it more illustrates the out-of-band performance of the subwoofer than the in-band performance).

However, it occurred to me I might have left some polarity change setting set in the left subwoofer because I have to crawl behind equipment and on top of it to read the settings, which is hard to do and I sometimes forget.

So, I did just that, bringing a mirror with me to read all the electronically displayed settings of the left channel PB13 Ultra.  I find that once you start adjusting or reading a setting, the only way to get back to the menu is to just let go of the knob, let the display time out after a few seconds, and then press the knob again.  Even just letting go, the adjustments you have made remain selected.  If you hold the knob down, the sub turns off instead of returning to the main menu.  I find this counterintuitive but finally figured this little detail out.

Well, it turned out I had set a 15 degree phase change in just the left subwoofer (it was zero degrees phase change on the right subwoofer).  That was not intended, or at least I did not remember it.  I readjusted the phase change back to zero.  This seemed to flatten the left channel response around 100 Hz.  But when playing pink noise in stereo again, it didn't seem to help the 100 Hz suckout much.

Looking at the right channel by itself (and I should keep reminding myself to only do adjustments looking at one channel at a time for the bass) I found the bass had been turned way down.  It had been set to -20dB at the PB13 control panel.  I flattened the bass out by raising it to -16dB.  But when I do that, this channel definitely needs a 100 Hz EQ notch with at least -1.5dB to have flat response, previously I had that notch as big as -6dB.

The current situation at 100 Hz is this: the response in either channel and with uncorrelated pink noise in both channels in flat.  But when playing correlated pink noise (centered) there is still a notch.  This is still a puzzle.  I usually adjust bass using the uncorrelated pink noise finding the correlated pink noise is always tricky.  But in some ways, and especially the bass, the correlated pink noise is more like real music.

Despite not fully fixing the 100 Hz dip (it does seem a bit better) actually many other aspects of the bass are now fixed and obviously better.  (For example, the right channel was very lean in the bass, and the left channel had a notch and other issues caused by unintended delay.)

Actually, maybe I dialed in the 15 degree delay at some time in the past to try to fix the 100 Hz dip in stereo.  But it doesn't.

Future adjustments may be more productive because I've now figured out how to adjust the delay separately in each channel of the Behringer 2496 DEQ.   The trick is you have to hold down the "B" button next to the Left/Right switch on the delay page to get it to shift from Left+Right Together to Left or Right Independently.  Then you can tap it to adjust only the left or right channel.  Not being able to do channel independent adjustments has been a huge problem for 5 years since I replaced the DCX's with DEQ's.  It took a careful re-reading of the manual to figure this out, it would have been more intuitive if the independent adjustment were the default but it appears not to be.  Behringer considers it a feature you can adjust delays independent regardless of stereo-link or dual-mono modes which normally control how the Left/Right switch works on other pages.  But that also means you have to toggle the Right/Left switch between together and independent modes, just for the delay page.

I also adjusted the 45 Hz notches separately for right and left subs.  They very much need to be adjusted separately, the left channel requires notching at 45 Hz but strangely the right channel doesn't seem to need it anymore, and I wonder if my Etrap in the far right corner of the hallway helps that.  Notching 45Hz when a notch isn't needed adds to undesired depression at 56 Hz.

By the afternoon of January 14, I had gotten impressively flat uncorrelated stereo response (btw, it's flatter than the scale on the right suggests, divide those numbers by about 6 because they're not the kinds of numbers we are used to, they are sixth-octave-bin-weighted dB's, I know this now as I can "move" the graph by an apparent 3-6dB or so with only a 1dB adjustment on my EQ).

This is also showing my new EQ's in the highs I worked out this morning.  I've progressed from one "Gundry-Linkwitz" dip to two tailored dips also helping with Acoustat resonance around 6kHz, to now 3 tailored dips in the 2-9kHz region, yielding very smooth and flat response.  As I added a new upper notch, I found I had to strengthen the other notches or else they seemed to bulge up.  In the end, these are the notches (as of Jan 14 at 1:41 CST):

9037 Hz, 1 octave, -2.5dB
5200 Hz, 1/2 octave, -4.5dB
2729 Hz, 1/3 octave, -2.0dB

I tuned the 9037 notch entirely by looking at pink noise on the RTA.  When I set the bandwidth less an one octave, it was clear that there was bulging around the edges of the notch.  Only with 1 octave bandwidth, and centered at about 9kHz, did I get essential flatness.  It might even look slightly flatter at -3dB, but that made it sound dull, I thought, so I backed off to -2.5dB.

Normally I have used an log dial oscillator and ear to set notch frequencies and width along with rough level.  Level is then fine tuned with RTA and music listening.  That's my "basic PEQ method" and it's essential for the bass where very steep EQ's are generally required to compensate for room nodes.  For upper frequency issues, and especially wide bandwidths, the oscillator isn't much use and just using RTA might even be better.  It's certainly easier.  I should go back and double-check these adjustments with oscillator.  That's the final part of "my method."  Always going back and double checking, sometimes using a different method.

Only the people I know who are far removed from science seem to think such relentless double checking is not a good idea.  Cranks think you can simply take one perfect test or measurement and be done with tests and measurement.  Things don't work that way.  Where I've tended to fail is not in relentless repetition, but in establishing the narrative so I remember what I was most recently trying to do by what I did, and so on, so I invent false narratives sometimes and end up going around and around in circles before I get the correct narrative sorted out--if I ever do.  (I believe that's not uncommon among audiophiles, only most do this differently--because they depend on highly unreliable and improper listening tests, so poor judgments are initially made, only later to be reversed--if the audiophile is sufficiently open minded to even reverse an earlier bad judgement--many simply double down.)

Establishing the narrative correctly is what I'm trying to do with this blog, and hopefully better this year than in the past.  Having a notebook and using it correctly was always my weak spot in science class.  I had to work very hard to make things up that I hadn't written down at the time.

Here are the latest EQ settings as of Jan 14 2PM

Here is the Stereo-correlated pink noise response, showing a notch at 112 Hz (either it changed slightly or I imagined it wrong):

Stereo-Correlated Response

Otherwise, it looks pretty good, still showing a bit of peaking at 71 Hz (which I am suppressing with a 13dB notch which doesn't seem justified by the uncorrelated response).  The highs look very smooth.

I have dialed back the notches on both sides of the 112 Hz notch, and nothing seems to make much difference.  It varies greatly by room position, suggesting it is a center room cancellation mode, not something caused by other EQ's.

Right Channel

 As stated (I got something right for once) there is no "100 Hz" or nearby notch at 100 Hz in the Right Channel, in fact there is a slight peak at 100Hz (which is already being suppressed by EQ in this channel so it isn't far worse).  At the same time, there isn't ANY peak at 45 Hz, the frequency that has dogged me since I first got subwoofers.  The EQ for 45 Hz in the right channel is dialed back to zero.  This hasn't always been true in the past, I remember it being as high as -9dB.  It may be because of the ETrap in the far right corner of the hallway nearby (but I should test that idea).  This channel is not contributing directly (or uncorrelatedly) to the notch at 112Hz in the correlated response.

Left Channel

Nor is there a notch at 112 Hz in the Left Channel, however there is a tiny dip (remember, 6dB on these scales is really like 1dB) at 100 Hz.  I am not attenuating 100 Hz in this channel.  Conversely, there is a slight rise at 45dB in this channel, which would be far worse if I were not already attenuating it by 4dB.

Both channels need to be heavily attenuated at 40Hz, where there is still a rise showing.  But I do not attenuate below 40 Hz.  I used to even have difficulty getting a solid 32 Hz for some reason.  That appears not to be the case now (but the only thing I seem to have changed is the Acoustats, which aren't even playing at 32 Hz).  32 Hz is musically very important.  If I were to attenuate it, I'd weaken that musically critical region, and experience important regions below.  I think the bulge at 32 Hz is also position dependent also.  But mainly, it's new, I haven't investigated it, and I don't even think it's a problem yet.

Many Audiophiles use Room Curves ramping up to as much as 20dB below 100 Hz.  A tiny octave bulge around 32 Hz of 2dB or less is not only not a problem, it's possibly beneficial.  (Divide the scale numbers shown by 6 because we are considering the whole spectrum.)

Possibly the solid 32 Hz is because of the center room position.  My previous position, before the 2+2's, was about 1/3 from the front.

Home Sweet Home

My almost annual trip to California resulted in visiting only one currently active audiophile, the very industrious Roger who always has endless audio construction projects and modification experiments in process.  Well this time I thought his sound was far better than in a long time, and possibly due to his use of actual commercial amplifiers, Manleys, which looked sufficiently muscular and I'm sure were undeniably hifi.  Previously he was into SET's with a home made SET having an enormous tube.  SETs can have serious rolloff, and I think his might have.  However, Roger attributed the most recent improvements to damping, specifically the metallic 3M damping material he is now applying in over inch thick layers to potentiometers.  (This looks like great stuff BTW and I think I will start trying it myself in applications where hockey tape wouldn't work.)  He's tossed his transformer volume control (very complicated and expensive) because, in the end, he was unable to find a way to properly damp it (all those transformers!).  Still present were a few of his amazing huge clamps, (everything must be clamped rigid, Roger believes) but fewer clamps than before, or so it seemed, maybe I wasn't looking hard enough.

His system IMO has generally lacked high frequency bandwidth as it's first offense.  He's somewhat apologetic about that, claiming great hearing loss.  However, this time it was much more extended than previously I thought, despite his still taping off the electret tweeters on his Zu speakers.  It actually sounded tolerable bandwidth wise, also with new subwoofer.  (Perhaps the bass even makes the highs sound better.)  It even had one positive quality, solidity, which my system relatively lacks.  A lot of speakers based around cone speakers in small rooms have solidity, though it's often too much in-your-face solidity, Roger's was about just right, in that one parameter.

However, come home, and the improvement otherwise in my system is staggering.  I don't have a thin midrangy system like ESL-57's.  With my gigantic subs, electronic EQ, and huge Acoustat 2+2's, I still rate well on solidity, though it's highly dependent on the actual crossover, EQ, delay, and other adjustments.

And then soon my system became even better.  I was then so inspired to do some more oscillator sweeps, and 1/6 octave phone analyzer readings on the Stereophile Pink Noise track.

I clearly did show some loss at 100Hz in one channel.  Strangely there was NO notch at 100 Hz, (though I had set a notch to 104 at one time, the size was 0.0dB) just notches at 72.6 and 130 which I changed to 72.6 and 140, which somehow made a larger difference at 100Hz than you might expect, not flat but not terribly sucked out anymore.  This improved the "solidity" significantly.  Note the subwoofer is crossed over at 100 Hz, but notches as high as 140 make a significant difference to the sound.

I also fine tuned the Linkwitz range notches to 2729 Hz (1/3 octave, -1.5dB) and 5200 Hz (1/2 octave, -3.5dB).  These not only make for a smooth dip in the 2-3Khz range as various people have recommended, it flattens some issues with the Acoustats themselves, which actually seem to have a slight peak right where it hurts the ear the most around 6kHz and also around 2.7Khz  Previously I used a single notch around 3.5kHz, but that left lumps at the new notch frequencies.  I'm not sure when I zeroed out the sub 1000 Hz notches in the midrange.  When I got the 2+2's at first those EQ's, which I had tuned using the 1+1's,  seemed OK, but later it seemed they didn't quite fit where notches might be useful, and they weren't as much needed.  I didn't photograph the midrange EQ in November when I did my first official 2+2 sweeps and adjustments.  I think I might have zeroed out the sub 1000 Hz notches then, and/or added two notches above 2 khz, but they weren't exactly the same as now, or I added different sub 1000 Hz notches.  Anyway, in the interests of restoring solidity, and perhaps more, I've removed all sub 1000 Hz notches for now from the midrange, and have once or twice tweaked two notches above 2kHz (there could possibly usefully be a third around 8-10 kHz, just to flatten/smooth the response, but that "artistry" will require a lot more development).

The sound even more improved, and more solid, I had accidentally moved the listening chair too far back during packing but lined up to the original tape which puts the ear about 6 inches beyond the peak of the roof, works great.  I started going through my favorite "special" discs, DVD-Audios and SACD, and then just kept up playing CD's mostly, my new years resolution being to play more music, and now I'm pretty much keeping it always playing, most recently from discs since they are easy to collect from my racks (not so easy to put back).  Each different method of selecting music (each different App, or physical media storage system, etc) makes you choose different music--a sadly uninvestigated phenomena.

I've also recently sweep tuned the kitchen subwoofer for greatly improved sound.  It turns out the room exaggerated 41.8Hz, where I now have a -15dB notch 1/2 octave wide.  With this change, FM no longer makes one sick near the sink, for example, because of throbbing bass.  I may now be able to switch back from the remarkably fine Sansui TU-D99X, thin but possibly their actual finest model which introduced the Walsh-type FM decoder they patented, to my still best tuner of all time, the Kenwood L-1000T.  The latter had become intolerable because of the bass exaggeration.

With these minor (though, possibly the most important) changes out of the way, I then moved on to Tweak-O-Mania!

Just before my trip I had been in crunch mode getting my house "ready" for the trip in various ways.  The biggest project of the year was making the old Kimball organ redundant by sampling it, and then getting it taken by Salvation Army.  But projects continued right up to the last minute, including fixing tile in the small bathroom, hoping to make my house perhaps just enough more liveable for someone to move in (which, wouldn't be expected, but there is always a remote possibility, and those possibilities are important if for nothing else than one's own self esteem--I promised to get rid of the Kimball 6 years ago for one thing, and I'd been promising to fix the tile for several years).

So anyway, I don't really mean to complain so much, but because of all the other stuff, highly visible audio project were being deliberately postponed.  If I barely have enough time for all the other stuff, which indeed I barely did, just barely completing everything I'd planned to do, I wouldn't have made if I had dilly dallied with audio projects of questionable importance.

BUT NOW I COULD DO THEM!!!

So finally I had time to fix the big speaker wire messes, including the ways that both supertweeters were hooked up with big extra coils of wire and stacks of connectors that would be hard to explain to another audiophile (but could be explained, if I only had the time, and a patient enough listener).

The supertweeters are now (20 days after returning home) hooked up pretty much correctly, with the wires from the amplifier running to "connector blocks" where they are merged with the thinner gauge wire that connects to the roll-off capacitors and both supertweeters.  The connector blocks are actually repurposed Furutech tightening bananas, which accept a huge gauge of input bare wire.  I strip and twist a generous portion of all the wires together and shove them all into the bare wire inlet of the Furutech and then tighten the two screws as much as possible.  They most connect to the two of 11 gauge wires for each polarity of the Canare 4S11 that runs up to the connector block, together with the 16 gauge wire stranded that runs to the capacitors and speaker terminals.  They do this easily.  This is one idea that I've sort of adopted from my friend George, who uses a special kind of copper ring for joining pairs of speaker wires by a similar method.  He claims his copper ring is the best, blah, blah.  But the same principle can be used with most anything that clamps onto wire, I decided, and one can hardly beat the finely gold plated copper used in the Furutech connectors.

Actually, I fear this clamping method isn't as perfect as George believes.  If there is any room for oxygen, the copper will corrode, and corroded bare copper is a semiconductor.*  The bare stranded wires are not being clamped tightly enough to eliminate copper corrosion.  It's better to have silver plated copper in the first place (George strongly believes that silver or silver plating causes harsh sound, but that's because his reference is the softened sound he expects despite his transducers), then clamped strongly or soldered in some way that doesn't permit air oxidation.  But as far as resistance, etc., the fairly large mass of wire now clamped with two screws in the Furutech connector is good enough for the supertweeters, I have decided.  And it neatens up the wires, gets blocks of connectors off the floor, and it makes the wires as short as possible, with a new twist I'm also employing in the midrange.  I use lower inductance and resistance wire (the Canare 4S11) for the longer runs, and the slightly higher inductance and resistance wire (my silver plated teflon coated milspec copper stranded wire) for the shorter side.  It just naturally happened I had already set things up this way for the super tweeters, but I had a coil of the teflon wire on one side connected to a block of connectors because I had never had time to get around to trimming the wire connections to fit (ever since I had to move the supertweeters inboard of the Acoustats after upgrading to the Acoustat 2+2's).

(*George used to use copper stranded speaker wire that had turned sea green because of corrosion.  He said it's the best just like that, determined by his listening to it.  I'm not sure what kind of wire he was using when he quit playing his stereo 3 years ago, but it's certainly not silver plated.  He generally likes to space wires as much as possible, which creates a huge amount of inductance.  This is all NOT high fidelity to me, but it works for him.  He says that what I do, and things I often say, don't make sense.)

In redoing the supertweeter speaker/amplifier wiring, I discovered that one supertweeter was formerly out of polarity, or at least so it seemed.  Everything is now visibly correct from the speakers to the amplifiers.

I also had a very ugly block of connectors for the right Acoustat, where connected a short piece of 4S11 to a short piece of 16G "audiophile" wire (using polypropylene inner sleeves protecting the bare copper with the corrosive and poor dielectric contact with the vinyl covering) that actually connected to the acoustats.  This ugly block of connectors sitting on the floor consisted of a pair of Furutech locking bananas, a second pair they were screwed into, and a third RadioShack Gold connector merely to keep the two pairs of pairs of Furutech connectors properly spaced.

This all became necessary when the original short piece of custom terminated 4S11 was no longer long enough, either because I have moved the 1+1's out from the wall more (which I started doing in spring of 2019) or because I switched to the 2+2's when then had to be moved as far as possible to the sides, with the supertweeters now on the inside because I can't fit them on the outside and still have a passageway through the room.  So I've had this ugly block of connectors on the floor for about 6 months or more and it's bugged me every day.

My first approach, however, didn't seem fully satisfactory.  I simply soldered 5 inch pieces of the teflon solid wire to the ends of the Canare 4S11.  This solder joint was made in a fairly convenient but questionable way.  I simply wrapped a loop of the silver coated solid wire around the stranded copper, squeezed it down as tightly as possible, and soldered with eutectic solder.  After the pressure of the iron was removed, the wire always slinked a bit, making me worry about cold solder.  I worked on this through numerous attempts for two hours, and finally decided it was good enough, the joint looked smooth all around, just not quite as shiny as I would have liked.  I then was unable to find my "rescue tape" and all my shrink wrap was too small, so I had to cover up the joints with vinyl electrical tape.

That got rid of the connector blocks, but it looked ugly to audiophile eyes and thinking.  It would be much better to use a single piece of wire, preferably the 10G solid core silver plated teflon coated hand twisted wire I use for most everything else now.  But I would only want to do this if I had a locking banana of the "straight" kind, and not the "bent" kind like all of my Furutechs (which I just seem to have run out of again, I buy them in bunches now).

Well, it turned out that in fixing one side of the supertweeter wiring, I freed up a short piece of wire that had exactly the right kind of "straight" locking banana I needed, AND it had locking screws.  (I have some other pairs, of terminated 4S11 with straight locking bananas, but the 4S11 wires are crimped into the bananas, not screwed.)

These straight locking bananas were just what I needed to take one existing short piece of teflon twisted pair wiring and put it to use for the right Acoustat.  Crawling around the floor and especially making the connections behind the highly modified QSC ABX box I use for amplifier switching still took almost two hours of work, but finally the right Acoustat was wired nicely by audiophile standards, with no electrical tape or ugly solder joints.

At first, however, it sounded harsh.  I had the wire almost 18 inches off the ground, going up and over the Acoustat interfaces and the stack of digital equipment down to the ABX box connectors.  It occurred to me this "high" run could be picking up distant interference, or perhaps vibration.

I lowered the wire by bending it (solid core wire must be bent to each new shape) down closer but still about 7 inches above the floor mostly.  I also "damped" the piece of wire by hanging a Velco strap tied to a loop of phone wire serving as a dumb weight, in the middle of the wire run.

At this point it sounded great, and the soundfield seemed more unified than ever before, or at least a long time.

I then remembered that by switching from 4S11 to 10G twisted pair I was approximately doubling the inductance and 50% increasing the resistance.  This would make this shorter run of wire from the ABX box more similar to the twice-as-long wire (now 4S11) running to the other speaker.  So, for the first time in a long time, perhaps, I have basically equal speaker wire inductance on both sides, and the resistance is fairly close also.

I've never been one of those people to axiomatically cut all speaker wires for stereo channels to the same length.  I know this has virtues, but it then leaves you with useless coils of wire, which may better pick up noise and have leakage characteristics.  And with all the wires I have, it gets maddening very quickly.  So instead I terminate each wire to the more-or-less exact length needed (usually with some extra margin, but over time, I've been finding margins not useful anyway, and I just cut things pretty close to the lengths needed).

So just 20 days back home, I've made major improvements some delayed perhaps by years, and I'm enjoying music more than ever.

The rattle in the right Acoustat has not been noticeable, even though I'm consistently playing at the "realistic" level (1.0V input max to the Hafler which is just below it's rated sensitivity for full power).