My first two days with ARTA

I can't really create anything like a time coherent speaker system either just by listening, or by back of the envelope calculations.  Nobody can.  It's too complex.

Everyone (except tweakophiles) agrees.  Measurements are required.  Every piece of the system has it's own time behavior, and crossovers and equalizers further modify that.

REW is really about room acoustics, and not so much speaker performance.  From the standpoint of REW, speaker perfomance is a given.

To design a speaker...which is effectively what I'm doing with my combination of subwoofer and supertweeter to a "full range" eelectrostat, I needed a speaker design program.

Many years ago, when I decided it was hopeless to try to improve on the LS3/5A, I was using Liberty Audiosuite 3.  That was a huge step up in my measurement capabilities, and served me well from 1999 to about 2010.

Using MLS (which I thought was a requirement--but now I see it was merely an optimization) and auto-magic gating (that really IS a requirement, unless you do outdoor or anechoic testing, or simple time recording instead of FFT convolution) LAUD3 was always able to construct plausible step response curves.

But that program HAD to run on an old fashioned ISA bus PC which could host a DSP soundcard like the Turtle Beach FIJI.  When those parts started to fail, I was out of luck, for a decade.

For the past 5 years I've been occasionally trying to get my LAUD3 system working again.  Without getting there yet.  There's always some issue that comes up in the setup process.  It wasn't much different in 1999, except the issues weren't so much hardware failures.  Getting the Fiji working, and re-working after every system upgrade, I figured I had re-installed Windows 95 about one hundred times.  It took re-installations because the Plug-n-Play system learned things the wrong way, and I had to start from scratch.

Anyway, I hold onto the past too much in some cases.  I should have moved on.  I did try to move on to the REW program, but it is unsatisfactory for the speaker time alignment.  I've written about that.  I needed a real "speaker designer's program" like LAUD, and for the past year or so I figured ARTA would be a good choice.

Finally, a few days ago, I got started by downloading ARTA.

For the first day, I was getting nowhere.  I have yet to figure out how to do the sharp gating required to see the leading edge from my speakers.  If you use the first 3 methods for deriving an impulse response, and from that a step response, you are convoluting a recording lasting several seconds.  Everything beyond the first few milliseconds, maybe as few as 4 ms, is compounded by reflections.  This is therefore a room/speaker analysis, and not a speaker analysis.  To do the time alignment, we really need to look at the first few milliseconds somehow.

And none of the menus I've looked at yet have anything to do with gating.

But I figured out an easier and much more intuitive way to do this.  Friends have mentioned the concept to me before, as much as 35 years ago.  A storage oscilloscope!

Around 1985 I picked up a surplus storage oscilloscope, but it was a dud, a really poor oscilloscope with a really poor storage feature tacked on, and I never had much time to mess with it anyway with a full time job and endless attempts to find a good girlfriend relationship.

Fortunately, ARTA has a "time recorder" feature which works just like a pretty good storage oscilloscope.  And with this, I can look at the actual behavior as time unfolds, not a convolution of the longer term behavior.  This is another area where I think the professionals are too caught up in their own magic.  Simpler is better, or at least simpler.

Even then, it seemed like I was getting nowhere.  I could measure the latency of the panels and the supertweeters.  But it was way different from I expected, the the supertweeter considerably delayed.  Attempts to remove delay from the supertweeter by dialing back the delay setting on the Behringer DEQ seemed to do nothing at all.  My first guess was that even in this supposedly time recording, with the loopback being taken from my audio interface itself, the ARTA program was doing some kind of additional automagic time shifting, just as had plagued me before when convolution was being done.

But it turned out, the problem was more in the DEQ.  I noticed I was changing the delay setting for MAIN, but the AUX setting was not changing.  So I started changing the delay setting for AUX, and now I was able to vary the delay.  It seems that the Digital Outputs are part of the AUX circuit, not the MAIN circuit.

With this change to my method, I was able to adjust the delays for panel and tweeter (on left channel) to match within about 0.02 ms, by measuring one at a time and adjusting.  There are some tiny undulations prior to the supertweeter step up signal which I'm assuming are digital artifacts.  Other than that there is finally zero ambiguity.  Finally.  I feel confident enough in this approach that I can now fearlessly pivot to using a different sampling rate in the supertweeter path, 96kHz vs 48kHz, and restore my system to virtual 96kHz operation (albeit only in the supertweeter path...but that's where it needs the 96kHz after all), and re-set the alignment by doing this again...it only takes a few minutes.  For that matter, I can now easily "roll" the DAC used in the midrange path (the panels), and likewise readjust the alignment in a few minutes.

But moving on to adjusting the subwoofer time alignment, at first I seemed to be back in the same old boat.  Changing the delay adjustments in the subwoofer path seemed, at first, to make no difference, and this time I started out adjusting the "AUX" path.  Only after I did a number of measurements, first moving the mike closer as a sanity check (moving mike closer DID have the expected effect), and flipping the MAIN/AUX switch a few times, did it seem finally to be working again.

Perhaps the answer isn't so much adjusting the main vs aux delay, but flipping the switch so it records the change???  I notice also that the AUX path, which includes the digital output, is selected to "be the same" as the MAIN path in an earlier I/O menu page.  However, that page doesn't show where the adjusted delay is applied.

For now, it still seems that the answer is to adjust the AUX delay, but possibly flipping back and forth between AUX and MAIN to be sure it "sticks."

But even with that little technical glitch fixed, the Subwoofer time response leaves me with a conundrum.

The subwoofer response begins with an 7.5 ms negative cycle before the positive cycle begins.  This is large enough for me to conclude it is NOT a digital artifact.

I've also done endless earlier adjustments of the subwoofer polarity, using a variety of methods, including a polarity checking app.  As far as I know, the subwoofer is In Polarity, it just responds with this little leading negative cycle.

So now the question is, do I line up the midrange response, which appears not to have such a leading negative cycle (or, if it does, it's so small it isn't clear on the default scale) with the part where the subwoofer response begins, at the beginning of the leading negative cycle?

Or, do I line up the midrange response with subwoofer curve where it's back to baseline and now moving positive?  Or, do I line up the midrange response with the subwoofer curve where it's at the bottom of the leading negative cycle and is starting to move up, even though it hasn't reached the zero axis yet?  Or something else entirely?

I suppose it would help to understand what this leading negative cycle is all about.  I suspect now, if I studied the midrange time response I'd see a similar thing, only very much smaller.  If that turns out to be true, it could be this leading negative portion is related to the high pass response of all speaker drivers.  (I suppose ionic and fan-based speakers could be an exception, but those are hardly worth mentioning due to other impracticabilities.)

Linkwitz talks about this inevitable high pass response, and where it is most important to deal with is in the subwoofer, because it creates a phase error in the crossover with the midrange that can be quite large.

The phase lead is 180 degrees at the natural (or unnatural) low frequency cutoff of the subwoofer, but in his example reduces to 23 degrees at the 100 Hz crossover (same crossover frequency as I am using).

That's because the phase lead falls as the frequency goes higher relative to the cutoff frequency.

But meanwhile, the lowpass being applied to the crossover for the subwoofer has phase lag, which is supposed to be 180 degrees at the crossover frequency of an LR4 crossover.

At that same crossover freuqncy, the LR4 highpass has 180 degrees of phase lag.  Thus both highpass and lowpass have gotten to the same point, but in different directions.  As we move either way from the crossover frequency, they remain in phase, as the leading goes up or down and the lagging does the exact reverse.

Linkwitz proposes either using an allpass filter (and that is what he does) or a time delay system (applicable to DSP type crossovers) to correct the phase shift caused by the woofer's high pass function.

But now it occurs to me, why isn't he concerned about the midrange high pass function?  That highpass function is closer to the crossover point, and it seems to me would contribute MORE relative phase error at the crossover frequency.

In fact, the ideal for ordinary crossovers (without going to FIR linear phase types) is precisely this: subwoofer and midrange having exactly the same inherent highpass function.  One way to fix the problem is to alter the subwoofer's highpass function to match the midrange highpass function.

In my case, the subwoofer has rolloff very close to 20Hz, whereas the electrostats roll off around 40Hz--somewhat extended by their own resonance and in my case the room resonance.

So, actually, despite the vast difference in appearance of the time output signal (because...the subwoofer goes far lower but not as high) the actual phase lead difference at the crossover frequency is fairly small.

I'm sure most of my friends would believe in lining up the first semblance of output, whether it's up or down, phase lead or lag.  One of my friends would always look at speakers claiming to be time aligned, and if the voice coils weren't lining up, he'd say it is a lie.  I'd always say there were probably other factors, but I never knew what those other factors were, until the one day, for just a few moments, I myself got to talk to Linkwitz at a trade show (Santa Monica, 1987 IIRC, or maybe it was San Francisco in 1997).  He told me more in that one minute than anyone else has in a lifetime.  I asked why I couldn't have phase linear crossovers and he told me I could, with FIR, but that had other issues (as is aptly described on a webpage now).  And about the voice coil lining up not being indentical with time alignment, that it was due to the high pass function of the drivers (as is aptly described on a webpage now).

So it would seem a first order approximation would be to line up the signals where they begin, regardless of direction.

AND then, to possibly adjust further, to correct the relative phase lead at the crossover frequency.

From what I know already, I can sort of predict what that will be like.  At the crossover frequency, ignoring the effect of the crossover itself, the panels will have MORE lag than the subwoofer because they are closer to their cutoff than the subwoofer is to it's.

(The disconnect between this observation and Linkwitz' focus on correcting the subwoofer lag, confuses me.)

If the subwoofer is lagging by 23 degrees, the panels might be lagging by 46 degrees.

Lagging more, means the panels should be corrected with less delay than would otherwise be required.

So after lining up the starting points (which will, btw, require adding much more delay to the panels, for some reason the subs simply have far more delay than I expected, possibly because internal digital  processing) I will then possibly have to reduce that additional delay, perhaps back to more or less where it was before (!?!)

What is 23 degrees at 100 Hz?  360 degrees is about 10ms, so my back of the envelope calculation says about .64 ms.

A workable process might be to do the first alignment based on first driver movement, then decrease panel delay to maximize output at the crossover frequency, up to about 2 ms, expecting the maximum to be around .64 ms.

And now, the miniDSPs are online too

In past 2 days I wired up, programmed, tested, and listened to the new miniDSP OpenDRC 2x2 boxes.

Using a new 4 AC outlet Powerbox from Cullen Cable (nice, all metal and heavy, with real industrial grade outlets, teflon coated twisted conductors in a shielded cable), which I waited for since March when I ordered it, arriving last week.  That is where the 3 minidsp power adapters plug in.  Rather than trying to rearrange everything in front of speakers to make room for one new powerbox, I simply placed the powerbox on the huge Panamax 1500 power conditiioning UPS that everything else, except amplifiers, connects to through 5 now 6 sub-strips like this Powerbox.  I had previously been using another of these Cullen Powerboxes for the Behringer DEQ's, the Tact DSP preamp, and the ABX switch.  I decided these big heavy metal power boxes are good for powering digital devices.

The "plugin" software from minidsp is poorly documented and quirky, but works, and thanks to reading blogs at miniDSP (which is in Hong Kong, btw) I'm beginning to get a better handle on it.

At this point, the miniDSPs are simply doing standard IIR crossovers just like you would do with analog devices, and as close as possible to the DSP crossovers I had previously dialed into the Behringer 2496 DEQ boxes (in which I have now turned off the crossover functions).  So the change so far is simply moving the same or similar crossover functions to a new set of DSP devices.  This should make little or no audible difference.

The minidsp Open DRC 2x2 plugins have IIR crossover function built in (for the FIR type, you need to generate the taps using an external program, then load the taps file into the plugin).  I'm doing the exact same crossovers and frequencies as I was doing before using my Behringer 2496 DEQ boxes.  With one exception, I'm using a "real" 48dB/octave Linkwitz-Riley high pass cutoff at 17022Hz for the supertweeters, rather than cascading 3 12dB/octave butterworth cutoffs as I was doing before (btw, a cascade of 2x 12dB/octave cutoffs forms one side of a LR4 crossover).  So he supertweeter high pass is now a bit steeper, 48dB/octave instead of 36dB/octave, and 6dB down at the crossover point instead of 9dB down.  I might have chosen 48dB/octave before but I didn't have that option on the Behringers.  Now, on the MiniDSPs, I don't think I have an option to cascade 3x 12dB cutoffs.  So their capabilities don't entirely overlap, but the new capabilities are the better ones.

My first listen to pink noise seemed like the bass was too loud and sounding particularly "bubbly" like an aquarium motor, and somehow especially annoying around 60Hz.  I had erroneously re-enabled one of the low frequency notches in the Behringer during the changeover, as I determined by checking photos I made in March.  Re-disabling that notch at 63 Hz helped a bit.  (Sometimes notches make pink noise sound more "tonal.")

But it turned out the main problem was that I had reduced the supertweeter level.  Thinking the difference between the old crossover with 9dB attenuation at 17khz, and the new one with 6dB attenuation, I removed an extra 6dB compensatory boost (thinking that wasn't such a good idea anyway).  But it turned out, to get the sound back to "normal" I had to dial that exact same 6dB boost back in (I tried other possibilities).  Previously with the Behringers I had found up to 9dB boost to be useful, but I had limited it to 6dB.  So now the 6dB is equivalent to the 9dB I had wanted to do before but didn't.

I also tried changing the high pass frequency, all the way from 14kHz up to 20khz.  Now and in all my "optimal" adjustments for several years I get a tiny 6dB notch at 16kHz.  But it turns out if I lower the 17022 even down to 17000 it messing things up bad, or if I move it up things get bad.  You would think making the highpass on the supertweeters lower would "fill in" the notch, but it does not work that way at all...the opposite seems to happen, the notch gets worse and turns into a general HF depression from 16kHz up.  And similar happens when I move the crossover frequency up.  There's obviously some weird cancellations going on here, the point I had already selected, 17022 Hz, is exactly the one that has the least cancellation.

So what's interesting here is that it is showing the exact same kind of sensitivity to crossover frequency as it was doing before, in spite of the changed crossover slope and different hardware.  Optimal crossover points are still the optimal crossover points.  I am interpreting this as meaning the new system is basically working the same as the old one, all around, which is what I wanted to see at this point.  I only want it to change significantly when I move to using FIR type filters.

To really nail down the high frequencies and how to do this supertweeter crossover better I will need much better measurements, etc.  1/6 octave RTA on my phone isn't good enough.  And time alignment which I have only approximated.

I listened to "Hanson, The Composer and His Orchestra" where Howard Hanson discusses each part of the orchestra before they play a bit.  It was on one of Harry Pearson's lists I came across when skimming old Absolute Sounds recently, and clipped out for reference.  I bought every disc on the short list, including the Hanson, and it is as worthwhile as HP said.

Somehow, it was sounding much better last night than two nights before on the old system.  Hanson's voice and the whole auditorium space around him and the instruments sounded much more real.  During that previous listening session, I discovered my chair had crept backwards about 5 inches and fixed it, but even after fixing the listening position, I didn't remember it sounding this good.

So it's possible something about the new setup, doing the same or almost the same crossovers in new boxes, is better than before.  Thankfully it doesn't at all seem to be worse.  So now I can move forwards and try different things.

BTW it is far easier to make changes on the Behringer, where you turn knobs and push buttons.  The fine control on the miniDSP must be done with an attached computer, and the highly quirky miniDSP opendrc "plug in".  (App would be better description, the fact that they call it 'plug in' confused me for years because I was expecting it to load onto the miniDSP unit.  But no, the software that actually runs on the miniDSP unit is called "firmware," the program that runs on your computer and downloads filter data to the device is called the "plug in.")  So my plan was and remains to keep both the miniDSP's and the Behringers online, using the miniDSP for crossover and FIR-type filters, and the Behringers for IIR filters and testing which is much easier.  With the Behringers I can just turn a knob and watch for changes in the RTA.  With the miniDSP you have to type in the number, and if you make a mistake the entire crossover could be lost (17000 might turn into 1700 and destroy my tweeters).  miniDSP recommends making adjustments "offline."  But if you disconnect and reconnect you need to go through another minute of "synchronization."  I found the best way is to simply mute the audio when making adjustments through the synchronized miniDSP plug in.  When I am sure it is correct I unmute the audio.  But with Behringer, I can safely just turn a knob, which is faster and easier when optimizing.

The optimal DSP box would be something more like the Behringer--with a built-in display screen, knobs, and buttons.  If the Behringer added FIR capability, and added a few other missing features (steeper cutoffs with adjustable Q, per channel phase invert) it would be way better than miniDSP and I wouldn't need them.

However, just like many times before, I could not easily remove the low pass crossover for the subwoofers from one of the Behringers.  Once they start doing a low pass, they never want to stop.  Even after power cycling, pressing the "Reset PEQ" button, and other tricks, the Behringer insisted on still applying that low pass filter.  I only got it to turn off by doing this: first use the "bypass" mode to disable the PEQ.  Then turn the crossover off.  Then turn the bypass mode off.  And maybe do all that twice.

Audio Party Setup

It now seems like ancient history and not all for the best reasons, but on the first week of March I hosted a very nice Audio Party.  A number of people from the local audio society RCAS attended, including at least two who've never made it to one of my parties before (this was the 4th party in 6 years of being a member).

A good number of very challenging recordings were played, and my system by all accounts handled them all well.  There was was much praise especially about the low well balanced low frequency sound and power handling, and respect for the integration of big dynamic speakers with electrostatics.

There was only the slightest comparative criticism, which I felt had some merit, about my system erring on the side of "smoothness" vs "detail."  Though generally I don't find it lacking in detail so much as dynamics, and I describe the sound of the Hafler 9300's in particular as "sweet and pure" rather than "smooth" per se.

The critic suggested the smoothness vs detail dichotomy was the fault of my Hafler MOSFET amplifier.  While that could be true, I pointed out later I have compared MOSFET and several bipolar amps in my system, and while I always believed they would differ in dynamics, I could rarely hear ANY difference at all in non-blind testing, often confused the two sonically, and never got above .5 percent accuracy in blind tests, demonstrating that probably there is little or no difference.  During the party, I even ran the Aragon 8008 BB for a minute or two, a fine vintage bipolar amplifier, mainly just to show that I could, while noting that I preferred the sweeter sound of the Hafler to the slightly more aggressive sound of the Aragon, however there too I had never reliably shown I could even hear a difference.

I have indeed long believed that Bipolar amps have stronger punchier dynamics (erring on the side of expanding rather than contracting dynamics), MOSFETS do the reverse, largely because of the positive vs negative temperature coefficients.  However, well designed amplifiers will mitigate the tendency of the raw devices so it is probably not perceptible.  Anyway, I used to pursue the high end in bipolar amplification (as I saw it) with the Krell FPB 300, and I was always very happy with the sound of that amplifier, and much less it's tendency to run hot and shut down and require service.  Because of their positive temperature coefficient, bipolar transistors are notoriously difficult to work with.  They require fancy protection circuits and work best with tons of feedback.  They are inevitably more expensive per watt.  To get the same purity in a bipolar amplifier as in the Hafler 9300's requires an all out effort like the FPB 300, or better yet probably, a Threshold 12e.  Nelson Pass himself quit designing and building bipolar amps after the Threshold 12e and moved on to MOSFETS without looking back.  This is not uncommon in the industry, though some people like Dan D'Agostino has not and probably never will abandon bipolars because he like the sound he gets from them, and his customers are willing to pay the high price for all the sophistication required.

Also electrostatic speakers tend to err on the side of being laid back dynamically while dynamic speakers tend to err on the side of exaggerating dynamics with resonances.  So there's that too, but I'm not willing to give up electrostatic speakers in the mid range, only in the bass it's impractical to get much out of electrostatic speakers.

But the issue of "lack of detail" even possibly "smearing" is exactly the issue I hope to address with more sophisticated crossover functions which I plan to implement with my miniDSP units soon.  I think that is more a speaker issue than an amplifier issue in my system now.

Anyway, partly a result of this criticism, and opportunity, I afterwards decided to upgrade the pedestrian 18 inch power cord (which was a 16g STJ gold plated audio accessory cord) on my Hafler with a new 12g power cord of about the same length from Cullen (who I'm also ordering a second Power Box from).  Of course, the mere fact that I pursued Bipolar amplification with great enthusiasm and expense for 10 years up until last year shows that I too had, and to some degree still have, an interest in Bipolar amps for their potentially better dynamics, which I have never shown that I can hear either.  But I also appreciate the simplicity, reliability, and what seems to me to be mostly purer sound from MOSFET amplifiers.  I'm not really in the income bracket anymore to pursue the high end in bipolar amplification anymore (my choice might be a Threshold 12e, fully refurbed of course).  Anything much less than that, and I'd be missing the smoothness and purity of my Haflter 9300's, I fear, and they are no slouch in dynamics or detail either IMO.  I might note that the Nelson Pass himself has pretty much completely abandoned bipolar amps since after the time of the Threshold 12e in the early 1980's.  But I think I should get my Electron Kinetics Eagle 2 amps refurbed...they were another miracle bipolar amp with distortion even lower than the Hafler.  I keep trying to remember to do that.  Maybe this year.  I think they would beat the Aragon sonically if not the Hafler as well.  And DETAIL is their strongest point, along with DYNAMICS.

All in all, I felt it was the by far best sound in any party I've hosted by far, and I believe that view was shared.  I think it's worthy to record in full detail the setup used to achieve this.  I had made many crucial adjustments in the weeks leading up to the party.  I endlessly tweaked and re-tweaked the parametric EQ settings to give the smoothest sounding sweeps, which always seemed to translate into the best sound.  But much is also just as it had been, with a few obvious mistakes even.  But I believe it must be recorded 100% before I feel safe about adjusting anything ever again.  So here goes:

Tweeter Crossover PEQ

The tweeter crossover is really a hack of hacks.  Basically, it's put as many high pass filters you can without making the leadup to the crossover too shallow.   Each 17022 Hz LC filter is 12dB/octave, so with 3 of them in series I achieve 36dB per octave.  Somehow doing 2 or 4 didn't work as well in limited tests.  I also used to have a slight makeup boost but here you can see I eliminated that by setting to zero.  In the background, the tweeters are connected through 0.47 uF caps, actually making it 42dB/octave, and that's not counting the drivers themselves and their acoustic loading.  Those caps used to be required to eliminate ripple hum from the long-in-need-of-refurb Parasound HCA-1000a amps, in fact I lowered the intended value from 0.47 from an originally planned 1uF which would have yielded 20kHz cutoff, and instead the capacitor/tweeter system is cutting off at 40kHz, which seems weird, but works well adding an extra 6dB boost for the octave 20khz to 40kHz, as well as limiting amplifier power from any conceivable signal.  The Dynaudio D21AF tweeters in front have 40kHz response and 600W power handling and 8 ohm impedance, the Vifa's in back I don't remember right now.

I haven't really done the impulse analysis or anything to tune this correctly, mainly I've used the 1/6 RTA on my phone to get adjusted fairly flat to 20khz...and I wish I could see what was happening above 20kHz but have never deployed the equipment to attempt that.  By themselves, the Acoustats are dropping like a rock above 18kHz, and while I can't hear pure 18kHz, I've always felt that fixing the extension made the sound nicer and more real.  Making up for a steep HF loss is a lot tricker than it sounds, you can't get there by fiddling with bandpass filters, you've got to do it for real with steep highpass for the super tweeters as I am doing here, unless you have some supertweeter gizmo with the crossover(s) built in.  My Elacs had that and I added my own extra crossovers anyway to get it out of the mid-highs.

This needs to be completely reanalyzed and redone when I'm using miniDSP's.

Supertweeter Delay

I've dialed in 7.58 ms delay on the tweeters.  Note that I delay every driver so I can make changes easily, so this number is only meaningful relative to the delays used on other drivers.  The original plan many years ago was to delay the panels (closest to ear) by 10ms, then everything else less.  However everything has changed since then, now the tweeters are closer than the panels.  The current settings have resulted from guesswork changes ever since my last formal "time alignment" which I wrote about here, and even that was very ambiguous and probably wrong, but not bad.

Supertweeter Gain Offset

I've dialed in 5dB of "gain offset" boost to the supertweeters because all of the high pass filters result in many dB of attenuation in the passband.  Including the capacitors which cause 6dB of rolloff at 20kHz by themselves.  Above 20kHz I don't know whats going on, there might be a peak, but many famous speaker drivers have peaks above 20kHz and nobody seems to care.  My cloth dome tweeters have essentially no peak above 20khz or below either and I generally think various rolloff phenomena  minimize peaking above 20kHz, including the vanishing dispersion.

Right Subwoofer PEQ

Now finally to the tricky business, the subwoofer crossover and EQ, managed by 10 or fewer Programmable EQ's (PEQ's).  Currently I use 8 PEQ's for the Right channel, including the two 100 Hz High Cut filters that form the 24db/Octave Linkwitz Riley crossover rolloff.  The other EQ's were determined by laborious oscillator sweeping and years of evolution.  Whenever I hear something bad in the sweep, I try to fix it with EQ.  They may look weird, but remove any one and it gets worse.  There might be a better optimization, but I know this works pretty well.  And my philosophy remains to equalize the "deep" resonances that develop over many milliseconds.  Those are what you hear when slowly sweeping an oscillator.  When using short duration FFT measurements, you are only equalizing the "shallow" resonances.  When I run FFT's I never see the deep resonances I find by sweeping, and EQ's derived from FFT programs like REW look just as weird as mine if not weirder.

Left Subwoofer PEQ

I used even more PEQ's in the Left Channel: all 10 available.  It's in 2-ports-open mode which makes the response more tricky, and the corner position adds to that.  This is still tiny compared to the number of "taps" used by people who use FIR response correction systems...often with as many as 6000 taps which would be the equivalent of 3000 second order IIR's.

Left Subwoofer GEQ

Right Subwoofer GEQ

The tiny adjustment in the Left Channel Graphic Equalizer was a mistake and an oversight.  At various times in the past I've used the Graphic Equalizer (GEQ) to make additional response changes when I run out of PEQ's, or just because it seems easier in some cases.  Most recently I was fearing (incorrectly, I believe now) that the crossover was running out of steam cutting off the subwoofer above about 500 Hz, so I dialed in GEQ cuts at all frequencies above 500 Hz.  I later found these unnecessary and possibly harmful, so I removed them all, or tried to, but mistakenly failed to remove few half db adjustments in one channel.  After taking these pictures for completeness purposes, I dialed the GEQ for both channels back to flat, making the GEQ light on the Behringer DEQ 2496 turn off.  I don't believe these adjustments were helpful, but the historical society might need them for a fully accurate replay sometime in the infinite future.

Left Subwoofer DEQ Page 1

Left Subwoofer DEQ Page 2

Left Subwoofer DEQ Page 3

Now we get into the really black arts.  If you believe in linear systems, you should never use compression, but if you want a system that works well even when you throw "Bass Test" recordings at it, you may need Dynamic EQ.   Because I use the 2-ports-open option on the left subwoofer, it's especially vulnerable to chuffing and other high excursion noises below 25 Hz.  So what this does is apply a gradually increasing rolloff below 25 Hz as the level starts getting too obscene.  I would have thought the SVS subwoofer would do this automatically.  A professionally made subwoofer HAS to have some level of built in electronic compression or excursion limiting to prevent endless warranty returns.  However SVS has set their limits apparently far beyond where I would set them and apparently more to prevent physical damage than just bad sound.  And much of the bad sound may also result from room specific resonances and rattles too.

I adjusted these pretty much by seat of pants while playing some bass test recordings.  Basically, when it sounded awful, I adjusted until it sounded OK, but didn't compress stuff at normal levels (the Behringer displays show when it is compressing).  One problem is that the level settings here depend on many other things, including the level settings on the subwoofers themselves.  Every time I adjust other level settings, I should have remembered to also adjust the DEQ (and DYN) settings accordingly.  But most of the time, I have forgotten to do that, only playing catch up from vague memories months later.  So I don't really know how well calibrated these DEQ settings are anymore, and I can't exactly remember how I set them in the first place.  But all seems to work as intended pretty well, so they can't be too bad either.  Another thing that needs to be completely rethought and redone, when I have time, which may not be soon.

Right Subwoofer DEQ Page 1

Right Subwoofer DEQ Page 2

Right Subwoofer DEQ Page 3

The right subwoofer is in full "Sealed" mode, so it doesn't need quite as aggressive DEQ limiting as the left subwoofer.  But a little bit still seemed to help.

Left Subwoofer DYN Page 1

Left Subwoofer DYN Page 2

Right Subwoofer DYN Page 1

Right Subwoofer DYN Page 2

Now talk about black arts, in addtion to DEQ I also use DYN, pure dynamic limiting (aka compression) with no "filter" aspect, on the subwoofers.  If I really knew what I was doing, I might get away with only DEQ which I think is less sonically intrusive.  But I also use DYN because I couldn't quite seem to do enough with DEQ.  The intent is first to do DEQ, but if DEQ isn't enough, then I also do DYN.  But you may notice the threshold settings on the left channel are only about 2dB different, so there isn't much "space" where you are doing DEQ without DYN on my left sub anyway.  Except I don't quite understand if the DEQ is frequency selective both in filtering AND in threshold setting.  I need to check these out better someday.  One problem is that my adjustments aren't usually well documented, so there's no "milestone" setting to return to if I mess things up.  The main purpose of this blog post is to document everything so I can get on to more experimentation with less fear of being without a net.

Subwoofer Delays

The subwoofer delays were set by a quasi-measurement method I described in this blog about a year. Not long after that, as often happens, everything changed.  I replaced the Acoustat 1+1's with 2+2's and changed all the listening geometries all over again to get the best out of the 2+2's.  Because of the added width of the 2+2's, I had to move everything slightly to the right in my system to allow a passage through the front door.  This means I really did need to set the delays all over again.  Instead, I tweaked them, mostly by guesswork, and ultimately a few months ago by listing to some test signal in which I didn't really know "which way" was right, but I picked one way and it does seem to work pretty well.

Back when I was setting the subwoofer delays using an REW measurement I did not know that I could set the right and left delays differently.  Years ago, the left and right delays needed to be different by at least 1ms or so.  Now the spacing is more similar on both sides, and it doesn't need as much (or any!) difference.  However I never even tried setting the sub delays different primarily because I didn't realize I could do it.  Then I read online about one button press on the Utility screen that makes it possible to set the delays differently.  Since discovering that, I haven't yet taken advantage of it.

Subwoofer Gain Offset: 0

I have set the subwoofer gain offset to zero.  It's best zero here because you CAN get full level bass signals from time to time.  However, I do now have one small positive PEQ in one channel.  For awhile I was trying to be "safe" by setting the gain offset to -2dB to compensate for the 2dB EQ boost around 90Hz in one channel.  However I ultimately decided I didn't like this "safety" cut and removed it, thence readjusting the subwoofer levels to compensate.  I have yet to hear 90Hz clipping around the boost--where there are also several cuts active so the point may be moot anyway.  Actually, if it got that high for very long, the DEQ and/or DYN would kick in anyway.

EQ for Acoustats

Now finally here are the PEQ's for the "midrange" from 100Hz to 18kHz or so.  The essential part are the two highpass (LC) filters at 100Hz which form the Linkwitz Riley 24dB/octave crossover at 100 Hz.  I've also added a 12dB rolloff at 20kHz to help crossover to the supertweeters.  The Acoustats naturally drop like a rock at 18kHz, but things still work best if I roll off the electrical signal as well.  I've tried many ways of doing this, for the longest time I simply had Linkwitz Riley crossover for the panels and the supertweeters at 20kHz.  But that didn't give the flattest response.  Much tinkering later I ended up with a single HC on the panels at 20kHz seems to combine with the natural Acoustat rolloff to complement the combined highpass on the super tweeters for fairly flat response at least to 20kHz, and it does no good for any amplifier to attempt to reproduce above 20kHz into the Acoustat capacitive load.  It sounds best to roll off the Acoustat signal on all amplifiers I've tried.

Another thing worth noting here is the "Gundry Dip" I have engineered between about 2000 Hz and 6000Hz with two PEQ cuts which are actually centered at 2729 Hz and 5200 Hz.   Could also be called a Linkwitz Dip because Linkwitz explained why it is useful, and I do strongly believe it is useful.  Basically we get too much reverberant response 2-6kHz to sound natural in small rooms.  However, even if I didn't want a Linkwitz Dip, the Acoustats themselves seem to need almost this much correction in the 2-6kHz region anyway, they have some annoying peaks here, and that explains the center frequency choices.  The additional cut at 9037 was added for that reason--the Acoustats seemed to have a peak there that needed tamping down.  The result of all these EQ's is like flat response with a very slight dip of 1.5dB centered between 2 and 6 kHz.

And one more thing, the narrow cut at 119 Hz is to eliminate a  particularly bad resonance primarily in the right channel which may be room related (I think it's a back wall reflection plus subwoofer output thing).  It wasn't as much needed on the left channel but I wasn't willing to split the PEQ into two channels to separate them.  The ill effects if any on the left channel are greatly exceeded by the improvement on the right.  On both channels the subwoofer is still producing strong output at 119 Hz despite the nominal 8dB or so reduction from the crossover.

Delay for Acoustat Panels

All the other delays are really set referenced to the Acoustat delay, which was originally set at 10ms but gradually evolved away from 10ms because of other changes.  Now it's 7.3ms, which is less than the supertweeter delay but more than the subwoofer delay.

Right Subwoofer Tuning

Right Subwoofer Level

My Right Subwoofer is set up as a Sealed Subwoofer so I set the tuning to that mode also.  The level is set at -16dB with the level mode set to Normal (and not High Level, I worry if High Level means connecting to a speaker level signal), but possibly I should use less attenuation with the High Level setting, something I haven't investigated yet.  I tried to photograph the position of the input level switch, but from every photo I've taken the switch is so small you can't tell if its in the Normal or High Level setting.  Trust me, it was in the Normal setting as these pictures were taken.

Left Subwoofer Tune

Left Subwoofer Level

I have set up the left subwoofer with one port closed and two ports open.  This is a useful compromise between the quality of the sealed sub, and the raw output level of a ported sub and helps give my system just a little more punch in the deep bass (which is mostly monophonic anyway).  IIRC, SVS originally allowed for having 0, 1, 2, or 3 ports open.  For quite a long time I had one or both subs with 1 port open, but that doesn't work very well, 1 port is not really enough for this sub, it leads to chuffing.  The useful options now are 0, 2, or 3 ports open.  With 2 ports open you select the 16Hz cutoff frequency as I am now doing.

Supertweeter DAC at -1dB

Subwoofer DAC at -3dB

DAC for Hafler at -7dB

A complete setup description must also include the settings of the input level controls for each DAC.  The Subwoofer DAC is at -3dB, I could in principle lower the subwoofer input level controls and raise the subwoofer DAC to 0, but having the DAC at -3dB gives me some flexibility: I could boost bass up to 3dB without having to get behind both subs.  And the subwoofer level controls are already set way down.  I'm not sure why the SVS subs require so much input attenuation in my system, it might have to do with the THX recommended level for subs.

The Hafler requires a nominal 1.1V input for rated output, and my DACs put out 2V into unbalanced load.  Now in principle I could run the DACs flat out into the Hafler and have higher potential output level because the Hafler does have some transient output capability above rated output.  But I also have to think about how well my digital level space is mapping to the useful voltages on the Hafler.  Running the DAC flat out wastes about half the "information space" of the digital signal which would go to voltages that are basically never used.  So there's a tradeoff here, and I have set the DACs up so that when there is no digital attenuation (which I can do upstream at my Tact preamp) it's as loud as it can usefully be, but no louder.  Another factor is that I needed to balance the Hafler and the Aragon for ABX testing.  I originally set this up with the Aragon at zero (it requires way over 2V for rated output) and the Hafler at -6dB, which seemed like equal loudness.  But careful measurments revealed the Hafler was actually level matched at -7dB.  (Does this mean the Hafler is actually "more dynamic" than the aragon???).  So it is now set to -7dB, but if I'm willing to give up the ABX I could possibly usefully set it a dB or two higher.  But it probably wouldn't sound better, I can already play the Acoustats about as loud as they sound best, when they play louder they don't sound "bad," they simply sound mediocre.  But I don't want mediocre, I was out-of-this-world, which is what I get at the levels I can play with the current setup all the way up to maximum "0dB" attenuation.

The Ittok tonearm is set to 2.2g tracking force (the balance is correct for all the tape I have on the arm and everything) and the anti-skate is set to "maximum" just above 3g--which worked perfectly on a blank groove antiskating test record.  Strangely Linn fans have mostly believed the Ittok Anti Skate is too aggressive, so many simply turn it off.  I find the whole arm works poorly without anti-skate.  The antiskate seems to have an internal damping system which HELPS the arm/cartridge horizontal resonance.  That resonance went away as I increased the anti skating.  Perhaps that's why the Linn people often like turning the Ittok antiskating off--they hate all kinds of damping.  But for me, it was just the ticket to make this arm perfect for my Dynavector 17D3, along with about 2g of hockey tape on the main part of the arm tube (which you can see just the end of in the picture) and another 3g of hockey tape curled up just above the headshell which in combination lowered the arm cartridge resonances to just below 10Hz.  Those changes had very salutory effects on the sound IMO, giving it clean sound over all (instead of edgy and aggressive) along with very deep bass.

Components, from input to output:

Dynavector 17D3 cartridge (I like the extended and clean highs)

Linn Ittok LVII tonearm (with hockey tape modifications)

JA Michell black Delrin low-spindle clamp (basically made for Linn-like tables) with felt washer (trimmed)

Linn Sondek LP12 Valhalla (the Valhalla mod is basically a crystal controlled AC power supply for the motor, and this 1982 generation of Linns also comes with the Nirvana super soft tuned springs)

Oppo UDP 205 universal disk player, which plays discs and also serves as a high resolution Roon endpoint for my library and Tidal.

Denon DVD 9000 DVD player--I use this for HDCD because it is the best sounding HDCD player I've ever heard, based on the PMI 200 design I believe along with Denon's AL24 and dual differential Burr Brown 1704's.  The Oppo 205 doesn't do HDCD either, and HDCD is one of my favorite formats.  The DVD 9000 can also serve as an HDCD decoder because it has digital inputs, but I currently don't have that configured to process HDCD files on my harddrive because I permanently borrowed the Sonos ZP80 I was using for that purpose for another purpose in my house, and the replacement ZP80 I had in the closet didn't get a crucial update a few years ago and is currently not working as a result.

Roon, I should mention, and Tidal too.  I plan to try qobuz soon.

Sonos Connect, which serves my library and analog sources in other rooms in my house through Sonos, and as Roon endpoint, and also serves the tuner output to the rest of the house.

Pioneer F-26 FM tuner...one of THE best sounding tuners ever made, widely considered Pioneer's best.  It was their last flagship analog FM tuner, part of the high end Series 20 range (but without the
gaudy Series 20 logo) along with the relatively ordinary AM/FM F28.  It's far easier to make a great analog FM tuner than a great digital FM tuner...there are relatively few truly great digitals.  What makes this tuner great is fine internal shielding, the ultimate 7 gang Pioneer front end, and Pioneer's best ever balanced detectors.  Many believe Pioneer "lost it" when they went to digital tuners, none had the low distortion of their best earlier analogs, more like 20 times more, though the F-93 had useable fake stereo.  Yamaha, Kenwood, briefly Sansui, and Sony made great digital tuners, with Accuphase ultimately outclassing them all.  I'm still using the cheap looking (but incredible) Sansui TU-D99X in the kitchen.  I don't think it quite has the WOW of the Kenwood L-1000T (incredible soundstage, dynamics, transparency)  but it is always always pleasant and low noise--possibly even quieter than the Kenwood.  It is a very special tuner too: the very last top-of-the-line tuner from Sansui, one of the most legendary FM tuner makers ever, and it uses the prototype Walsh MPX decoder that was ahead of everyone else at the time (1984).  Walsh "or equivalent" MPX was long the standard using the Sanyo LA 3450 IC, but never as good as a true "Analog Multiplier" like the Kenwood L-1000T or L-02T or Yamaha T-85.  However, Sansui's version of a Walsh decoder (which they published in JAES about) was so close to the "Analog Multipler" sound, it's hardly worth quibbling, I think it must be way better than the 3450...it doesn't seem to have as much "information lost" sound as the 3450 sometimes has (though the Sony 730ES, which uses a 3450, also sounds quite nice).  The F-26 has a relatively old fashioned Pioneer branded MPX chip similar to that in other top Pioneer analog tuners, but the front end is so superb it still works great.

Emotiva XSP-1 Preamp: about as low noise and distortion as anything available today.  The phono is incredibly quiet even with a low output moving coil cartridge like my Dynavector 17D3 I get zero audible noise even with ears to the speaker at max gain.  With the 0.5dB resolution digital level adjustment I can optimize the level going into my Analog to Digital converter repeatably--there is an optimal setting for every recording so this is helpful.

Lavry AD10 professional Analog to Digital converter.  In my system all analog sources like Vinyl, and digital sources where the digital output is "downrezzed" like DVD-Audio, or where there is no truly representative PCM such as SACD or even HDCD, all have to be converted to PCM in my system to be processed by my digital equalizers.  I believe and find this digital conversion to be about as transparent as the best preamps, which is also what the noise and distortion measurements would lead you to believe.  The Lavry AD10 does an excellent job and has a very useful input level control AND level display that helps keep everything optimal.  For many years I even believed CD digital sources sounded better going through this extra conversion, but in more recent investigations I reversed this opinion.  What always sounds best is to get the full resolution signal, and if the digital available has the full resolution then the digital signal is preferred, but otherwise not.

Tact RCS 2.0.  I use the Tact "Digital Preamp" as a glorified digital input selector and digital attenuator.  I do not use the Room Correction features.

Three Behringer DEQ 2496 Ultracurve Pro digital equalizers, with the settings described above.

Four Emotive Stealth DC-1 DAC's: (1) supertweeter, (2) subwoofer, (3) Hafler 9300, (4) Aragon 8008 BB.

Hafler 9300 (refurbed) the best sounding amplifier I've had at home.  Designed by Jim Strickland, who also designed the Acoustats.  After many amplifiers failed to drive the Acoustats without blowing up, Strickland decided in the early 1980's to design his own amplifier circuit, which he dubbed trans nova, which would drive anything with total reliability.  The Hafler 9300 is nearly the last generation of this design.  David Hafler bought the Acoustat Company mainly so he could get his hands on Trans Nova. However, it didn't save the Hafler company so much as give them a good buyout price from Rockford Fosgate, who also wanted the Trans Nova.  As implemented in the Hafler 9300, the trans nova is about as simple as a push-pull amplifier could be.  It is fully DC coupled with no series capacitor and yet requires no servo, relay, or protection circuit.  It comes on instantly without any kind of thump, pop, or click.  No inductors either, and the MOSFETS run at a fairly nice high bias.  You might not be surprised if I told you that the Krell FPB 300 is a low distortion amplifier, but within it's power range the Hafler is just as good.  And the Hafler has seemingly endless peak power too.  Needless to say, the 9300 is unperturbed by any load, eventually you might blow the fuse, but I never have, even in blasting competitions with the FPB 300.  With it's tendency to shut down at the slightest excuse, it's the 110 pound Krell that is the paper tiger.

Aragon 8008 BB an incredibly reliable and powerful bipolar amplifier of considerable merit, nearly the equivalent of a 90's Krell Amplifier like the FPB 300, especially after some bias readjustment by me.  This is my current "B" amplifier, for backup and comparison.

ATI 1502 amplifier (for the super tweeters).  A quality amplifier, with neglible hum despite age.

QSC ABX for switching the amplifiers.  This was modified by me for best sound, the relay snubbers (capacitors to ground) were removed so that there is nothing in the audio circuit except very heavy duty relays and short heavy wire.  There is now galvanic isolation from the speaker connections to anything in the ABX box itself.  The risk is that at some point I might burn out the relays from high power switching without the snubbers, but there is not the slightest evidence that any wear has occurred yet...these relays were spec'd for switching 600W amplifiers like the pro amps that QSC makes.  They are bigger relays than I've seen before in any audio device.

SVS PB 13 Ultra subs, with Sledge Amplifiers.

Acoustat 2+2's.

Supertweeters consisting of Dynaudio D21AF's facing forwards, and Vifa tweeters facing backwards.   Both tweeters are "mounted" to highly damped wood boxes which were originally Rogers LS 3/5A speakers which I started mucking wih long ago.  The supertweeters sit atop 40 inch sand filled extra heavy Target (UK) stands.

Henry Engineering AES digital splitters (during the party, I was using two of these, one to split the digital signal for the 3 Behringers, and one to split the midrange signal for the two DACs that drive the Hafler and Aragon amplifiers).

Cullen Power Box 6 hospital grade outlets in a metal box with excellently well shielded AC cord, for powering all the digital devices.

Oyaide Power Bar for powering the two midrange DACs and the second AES digital splitter.  Because apparently the Stealth DACs might not be as well AC isolated as they should be, I had to put the AES digital splitter on the same AC circuit as the DACs to eliminate hum, even using balanced AES cables.  But now there is total silence.

Wiremold Hospital Grade Power Strips (the $100+ version with 15 foot cords) I use 3 of these to power all the analog and digital equipment on my side rack.  These grip plugs astonishingly well, often hard to get the plugs pulled out.  Because of the 15 foot cords, I do not have to use extension cords anywhere in my system, and all my power strips have hardwired cords (I hate power strips with detachable cords...one more point of common failure and/or deterioration).

Panamax 1500 power conditioner and UPS (for all non-power equipment).

3 AC circuits: 1 for the Panamax, 1 for the subwoofers, and 1 for the Hafler and Aragon amplifiers.  All are run with 10g wire and isolated grounding on the same AC phase from my electrical box.  The outlets are all Pass & Seymour 5242-AI "Spec" grade (aka Industrial).  I fell in love with these outlets many years ago when I mistook them for the very similar Pass & Seymore 5242-I outlets recommended by Bob Crump, the parts expert who collaborated with John Curl on the experimental amplifiers which led to the Parasound JC-1.  Crump liked P&S industrial outlets for the same reason I do: they have unparalleled grip.  Even the impressive looking Hubbel industrial grade outlets can't hold heavy AC cables straight out of the wall, they immediately sag as much as 20 degrees.  In contrast, the slip in the P&S is nearly invisible at first.  And you can really tell the difference when you plug things in and out too.  It takes real effort with the 5242-AI and almost as much with the 5242-I.  Now also it turns out my 5242-AI is a far nicer looking outlet than the 5242-I.  It's heavier and the cheap looking aluminum back frame of the -I version is replaced by the most robust brass you've ever seen on any outlet.  The AC connections on the side have a heavy brass "carrier" enabling the electrical wire to be wrapped around both terminals and through the carrier for ultimate contact.  Why did Bob Crump not recommend these?  Perhaps they weren't commonly available then--he doesn't even mention them in a classic online post.  I once used Oyadie purple outlets on my then only dedicated circuit, but they had about the least grip of all I felt.  I tried to use a PS audio outlet in 2010 but there was some issue either with the screws or the wire inlets and the electrician could not use them.  All my electrical work is done by a favorite licensed electrician.

All of my SPDIF cables are made by Blue Jeans Cable with Belden "Brilliance" 1694A coax.  I also use Blue Jeans LC-1 low capacitance audio cables for nearly all unbalanced audio connections.

Balanced AES cables: a variety from Canare, Mogami, Geistnote, and Gotham (some of the best pro brands).

Balanced Audio Cables: I've been moving from Blue Jeans to either more professional grade or more exotic.  I still use a Blue Jeans made cable with Belden wire on one subwoofer, but use Mogami Gold on the other subwoofer.  For the most crucial links between Oppo player and Emotiva preamp and thence from preamp to Lavry ADC, I now use balanced cables based on milspec silver plated copper with teflon dielectric: SurfCables Pro-2 Silver XLR.

Speaker Cables:  Hand-twisted milspec silver coated 12g solid core wire with Teflon insulation, also one instance of Canare 4S11  for the left side speaker run to balance the resistance and inductance on the right side which is half as long, and 4S11 on the supertweeter cables also.  4S11 has half the mutual inductance and half the resistance per foot compared with my hand-twisted milspec wire, and though it doesn't have Teflon dielectric it uses polyethylene, which is so close to Teflon it's hardly worth quibbling.  In fact, polyethylene is about equivalent and possibly superior in some ways to FEP, also called Teflon, used in many super exotic audio cables.  In fact it's hard to tell which products use what I consider "the real Teflon" (PTFE) and not FEP.  I'm not 100% sure about my milspec balanced audio cables...they might be using FEP.  But once again, so close, hardly worth quibbling, just that PTFE is the ultimate best dielectric next to vacuum, and FEP and Polyethylene are at the next rung down.  I have mostly tended to make speaker cables "as short as possible" and not necessarily "equal lenght on both channels" because the latter often requires a coil of speaker cable somewhere, which I don't think is good either.  However now I have a pretty good way of equalizing the cable parameters on both sides even if the distances differ by a factor of two: using a 4-wire (4 quad) speaker cable instead of 2 with each conductor being equivalent.  Actually the evolution was replacing the 4S11 on the shorter side with my hand-twisted cable (having twice the resistance and twice the inductance per foot)...previously the sides were unbalanced because I used 4S11 on both sides but unequal lengths.  I felt the improvement in imaging from having the cable parameters, if not the cables themselves, equivalent on both sides, something I've rarely done, last year.  The ultimate best using this approach would be to twist my own 4-conductor speaker cables using my milspec teflon wire for the left side.  That would require some time to make well.  Speaker cable capacitance is basically irrelevant in my system even more than most: the load is capacitive.  Actually it might work better to increase the cable capacitance...using 4-conductor on the short side and 8-conductor on the long side for example.  That might do something for "detail" too.

Cables are supported by Cable Elevators (tm) where useful to keep speaker wires separated from digital and balanced audio wires.  I also use DarkField mini cable elevators to avoid having cables atop equipment or AC cables.

Don't use 95% Ethanol for cleaning PVC

In unfortunate haste, I tried to clean some of the smell of the big plastic topped rotary encoder knob on my newly acquired HP 3582A Spectrum Analyzer (a standard in audio analysis).  Sniffing around the instrument, much of the smell seemed to be coming from this knob, possibly because of it's flat plastic surface being a sponge for odors like cigarette smoke.

I used Everclear (95% ethanol) and Q tips.  Fairly quickly it started creating streaks and even sandpaper like bumps in the PVC surface.  (It didn't do much for the smell either.)

I vaguely remembered many years ago hearing some electronics maven (probably an old tech at a surplus store) tell me that 91% Isopropyl was safer to use on plastic than 95% ethanol.

But quickly scanning through expert opinions on various sites, I was not finding anyone distinguishing the properties of ethanol and isopropyl on PVC.   Both could be used briefly but longer would cause streaking, read a typical opinion.

Finally, I found this authoritative table, which indeed shows that isopropyl is safer on PVC.  Ethanol is the WORST of the simple alcohols, gets a C rating.  Isopropyl is among the best of the alcohols, gets an A1 rating.  Butanol is A2.

https://www.calpaclab.com/pvc-polyvinyl-chloride-chemical-compatibility-chart/

I'm actually thinking the Pledge floor cleaner, designed for vinyl floors and based on Butanol and surfactants, might be the safest and effective.  It's supposed to be diluted 1/64-1/32 also, fwiw, which makes the Butanol very dilute.

Some pledge polish for hard surfaces (put on a cloth first) seemed to remove the smell and replace it with nice lemony scent.  But the lemony scent wore off and the original smell returned.

Linear vs Minimum Phase Digital Filters

https://troll-audio.com/articles/linear-and-minimum-phase/

Wow, I had never thought that the "linear phase" filters used as CD brickwall filters are so called because they have no phase distortion!  All frequencies are delayed equally.  This requires an acausal filter to realize (which could be an analog system with delay lines).

The price paid for this magic of linear phase is the pre-ringing artifacts.  The pre-ringing is at ultrasonic frequencies and typically quite small.

Reverting to causal minimum phase filters get you no pre-ringing, but the post ringing is far larger, and the pulse itself is distorted because of phase shift before the cutoff point.  (Just one example is shown.  The phase distortion and post ringing can be played against each other in different ways.)

Audyssey Multi EQ Editor

My current kitchen surround system is finally sounding pretty nice, using the analog outputs of the Oppo BDP95, duplicating the surround sides to the backs with 10ms delay (simply because I'm using Behringer DEQ set to 0ms, it's minimum latency is 10ms).

Using my 2005 Yamaha 7 channel receiver with Hafler 9300 amplifier for the front speakers.

I was playing A Night At The Opera by Queen DVD-Audio, and the Brothers in Arms by Dire Straits DVD-Audio.  The Queen disc especially uses the surround channels with great artistic flair.

But if I were to get an actual up-to-date surround processor with all the latest processing, something I occasionally think about, I would want the Audyssey Multi EQ Editor which makes it easy to customize.  Here's an old list of processors that have this feature:

The following existing AV Receivers will support the MultEQ Editor: Denon AVR-X6300H, AVR-X4300H, AVR-X3300W, AVR-X2300W, AVR-X1300W, AVR-S930H, AVR-S730H, AVR-S920W, and AVR-S720W. Also, Marantz AV7703, SR7011, SR6011, SR5011, and NR1607.

So I need not spring for the extremely pricey Marantz 8805 to get this feature.

More importantly, I think, I've just purchased a pair of side surround speakers that can be mounted at ear level in the kitchen because they are so small: Gallo A'Diva SE's.  The back speakers will remain at the near-ceiling level they are now, because no room for more speakers there.  I'm not sure what I'll do with the current ceiling height side speakers.  They might be repurposed for front front-high-wide speakers and modified with my Infinity Walsh Tweeters and new woofers.