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.