Disc Players and Transports

I started using a network audio system in 2006, Sonos, and I still believe it is a fine system fundamentally.  I Sonos Connect nodes (originally called Zoneplayers) to provide coax SPDIF digital signals to systems in every room, mostly from my kitchen Mac's hard drive (a 1Tb Fusion drive), sometimes making connections from analog sources in different rooms.  All digital signals are uncompressed 44.1/16, which I've always believed to be fine and good.  John Atkinson measured remarkably low jitter from a Sonos based wireless system in 2006.  I actually eschew wireless and hard wire all my nodes with Cat-6a ethernet which permits attaching analog sources in 5 different rooms simultaneously for uncompressed forwarding.  All nodes are individually wired to a central gigabit switch.  I've often dreamed of DIY replacement for this, but it's almost unimaginable with my time constraints now.  Newer network audio systems Never (as far as I know) permit attaching analog sources in different rooms.  Sonos was the first and only to offer analog inputs and it was why I chose them instead of Squeezebox, though I missed the Squeezebox high rez capability.  And most new audiophile computer audio systems are astronomically expensive, like $20,000.  I bought Sonos Connect nodes like candy because they're only $395, small price for another room of fully 2-way connected audio.

It's very convenient, but has never played high resolution sources, so I maintained a collection of disc players to play high rez such as DVD-Audio discs (which are sometimes incredibly good), DAD/DVD-Video discs by Classic Audio, which are all fabulous, and SACD's, which I consider pop high rez (widely used if not the best system) but still indispensable.  And HDCD's too, which are all among the best recordings I have, and the more I know about it the more I still think HDCD wasn't a bad idea.

Are Disc Transports better than Computer-network systems?

Now I've even had my mind turned to the idea that disc players may be different and possibly better than computer network based systems.  In any case, it's an excuse to play ordinary CD's on my Denon DVD-9000, which is just about as fun as clicking on things on a computer screen, plus it has class (and weight!).  In most cases now I take the analog outputs from disc players and resample them to digital (for my crossovers and DSP) using a Lavry AD10.  So, FWIW, I am preserving the sound of the different players, to the degree that it can be done this way (and it is my belief that the 96kHz/24bit sampling of the Lavry is extremely good and transparent enough to do this).  I haven't seriously explored the "better sound" I get from playing discs as compared to network bits, but in a few cases I've noted the difference between taking the analog outputs and the digital outputs, and in almost every case I've preferred taking the analog.  I consider my Lavry resampling to be as good as the best preamps, and probably better than most preamps.  Digital sampling sounds impossible but has been incredibly well developed by our technological society.

As someone with collector tendency, however, somehow I wouldn't mind having EVERY classic disc player in the rack, so I could try them all and vary according to mood.  Unfortunately my latest rack was only just set up a month ago and already out of space.

Anyway, I've long felt I wanted a Real SACD player, and not a Universal not particularly noted as having the best sound on SACD's like my Denon DVD-5900.

I've long scanned eBay for SCD-1's and SCD-XA777ES's, the two original and very similar super high end SACD players by Sony.  To me, these represent REAL SACD, as Sony originally intended it.  I don't necessarily consider these players to be any sort of ultimate for regular CD's, after all Sony was primarily hoping to sell the idea that SACD's were superior to CD's, so one would imagine these players pulling out all the stops for SACD but possibly falling a little short on CD (and some reviews bear this out).  Some do say some later machines are actually better on SACD also, such as the much more cheaply built but still pricey XA-5400ES.  Somehow that latter machine has never appealed to me much.  Anyway, even if it did, I wouldn't want to spend that much as current used pricing for such a lightweight machine.  (I could be tempted to spend that much for a heavyweight universal, such as an Esoteric UX-3.)

I especially loved the balanced outputs of the SCD-1, which would be fabulous into my Lavry.  I also noted the especially clean noise floor of the SCD-1 in Stereophile review, and I attributed that in part to the balanced outputs (and multiple overbuilt power supplies a factor also).

But both those players had a very fancy but notoriously unreliable CD mechanism, the "fixed laser" mechanism.   There doesn't seem to be any real advantage in making the CD spindle move instead of moving the laser.  It just seems to be something rather difficult to accomplish, something that few companies would even attempt.  Sony of course did claim it was better, but there are many machines people believe to be better transports than the SCD-1.  To many observers, and to me, the "fixed laser" motor seems to be a marketing gimmick with no real advantages, and something that saddles the owner of these players with future repair costs.  Sony never seems to care about that, in fact their history has been one of a very high level of planned obsolescence, including not making stuff to last very well beyond the warranty period.  This is probably why Sony, which now seems to own just about everything, should be the most successful company ever but instead is barely keeping the lights on in their vast empire.  What has happened is that generation after generation of new customers has left Sony behind after being burned one time too many.  I've been among those burned many times, though I'd still been a big Sony customer until very recently (when they don't seem to make anything I want anymore).

Anyway, the 3rd Sony SACD player to have SACD electronics reportedly similar to the SCD-1 (said to be something like 75% the same) is the Sony DVP-9000ES.  There is hardly a time on eBay anymore when there aren't a half dozen or more of those for sale.  After years of scanning for them, I've gone ahead and purchased one with a newly replaced laser tested on multichannel SACD's--the hardest to play.  This cost more than the current average price but I hope will be worth it.

The DVP-9000ES has a specially enclosed but ultimately ordinary DVD/SACD/CD mechanism as far as I can tell.  The laser is shared with other units and plentifully available.  I don't even know if it's all that special but people have said the 9000ES is a decent transport...perhaps as good as the SCD-1.

This machine will be my go-to machine for SACD's.  For CD's I'll stick to the Denon DVD-9000, the differential 1704 based behemoth.  For DVD-Audio and DVD-Video-Audio I'll use the Denon DVD-9000, my current Oppo BDP-95, or perhaps get a new BDP-105 for the living room.

I would have preferred to do DVD-Audio on my DVD-9000, but that feature is broken on my unit now.  I could also repurpose my Onkyo RDV-1 as a DVD-Audio player and use something else, perhaps my Denon DVD-5000, as a supertweeter DAC.

Does a transport make any difference?  Well I don't know, but I imagine if the transport is temporally unstable to any degree, this is going to affect everything downstream to one degree or another.  I suppose instability could be erased simply by reading an entire song into memory.  FWIW, people I know had disliked memory-based players.  At the risk of sounding technically illiterate, I actually don't know if any electronic systems have less jitter.  I've very suspicious of all asynchrous digital systems like ethernet, and I have always thought USB to be subject to the same highly jittery phenomena but it might have a kind of "synchronous" mode, I don't understand that but I still think it's not fully synchronous.

One has to understand that even if things are buffered, reclocked, or whatever, at the end, those subsequent phases will ALWAYS have to adapt to the source...so the source matters.   (And a jittery transport layer matters also, though entire-song delivery would seem to be safe, so where is it unsafe, I don't know.)  It may be in principle that something based on a later clock can simply clock things out.  But if _it_ has to wait longer or shorter, that may cause some unintended small difference, say from the heat in the circuitry for having to wait longer, or negotiate with the source for more data packets more often.  So one can try to wall off noise in the time domain, but no such wall is ever perfect, computers don't really work on 1's and 0's, they work with electrons and semiconductor junctions, everything shifts a little, and for most work the absolute clock stability isn't important once you're off the chip and even perhaps there.

So I really don't know if semiconductor memory in a dedicated circuit, or any actual circuits, would abolish jitter any better than a spinning disc.  I think it probably would, but I don't actually know.  It could well be that if disc transports sound different than electronic sources, it's because the disc transports are expressing their character (of some kind) in the timing.  And in this case, where it isn't captured by conventional measurements or ideas, I'm all for this kind of difference.  I want to try all the different flavors of candy in the shop of my imagination!

So what looks good to me?  Well, the VRDS Neo players from Esoteric, I'd love to have UX-1 or UX-3 because those are Universal Disc players, and that's ultimately what I need.  Even if it isn't Sony, or even "real" DSD, it's close enough to be a Universal for me,  high end mechanism and quad 1704's.  These are the most heavyweight low resonance mechanism ever.

I'd love to have a CEC made belt transport, even in the Parasound 2000 version (which was reputedly based on CEC's cheapest model).  You can't argue with top load and screw down, that's always been the best way IMO.

Pioneer's Stable Platter machines look interesting, but only a few use "the real" Stable Platter with heavy parts.  These were incredibly high end machines I knew nothing about since I wasn't much following high end audio in the mid 1990's.

Sony did make a few excellent pieces of machinery in the 90's.  I believe the XA5ES, one of Lampizator's favorites, is one of these.  After that, and after the whiz bang SCD-1 and SCD-777ES, it's been pretty much all plastic.  I think the key assembly of the mechanism of my 9000ES is all plastic, if not the entire thing.  I recall my 507 is largely plastic.  (I fixed the 507 mechanism replacing the quickly broken rubber band with a weight.)

Phillips really did more in the direction of engineering things to last (but then, sometimes didn't).  Famously they made the CDM-1 mechanism and a large number of all metal derivatives.

The original Theta Data transport was based on a Phillips Laserdisc transport (basically, that's all it was, with thick panel added and tiny proprietary board added, which according to Lampizator, doesn't do anything useful, but he could be wrong, and price raised from $500 to $5000--US Engineering!).

Nevertheless, Lampizator has loved that transport and thinks there is magic in the much heavier mechanism and 30 times more powerful motor in a laser disc player.  So....there's a wide field of interesting looking laserdisc players.  Notably the top Pioneer players, that look to have excellent mechanism and digital output, but more research necessary.  Meanwhile I find the top Pioneer Stable Platter players, a cheaper version of which was used in a Wadia player, most interesting.  And I had not known at all about such things until now.

I do think that keeping the disc stable and clamped to a heavy disc on one side and tight clamp on the other is a most excellent idea.  I have long liked serious looking players with user tightened weights.  The SCD-1 appealed to me but now I see was not the giant I thought, but I still like the top load idea, Krell did some most excellent top load CD players and there may have been others. But it seems Pioneer was accomplishing this full disc clamping idea through a well engineered mechanism long ago, with just an ordinary (well, large) drawer.

For those with transport obsession (or is it transport lust?)  one essential guide is a transport list, this is a good one, though I think I've seen others that are about the same in comprehensiveness, and nothing is ever truly complete.

Every time I look at these lists I find interesting factoids.  For example, my Denon DVD-9000 (aka DVD-A1) uses a Hitachi HOP-1000.  That rings true as I know it needs a Hitachi laser.  The Esoteric X-01 and X-03 use variants of the VRDS-Neo and the UX models aren't even listed (but I think they use the corresponding transports).   The Onkyo RDV-1 uses a Mitsumi PVR-202T.  And, what I was just checking for today, the Parasound C/DP-1000, C/DP-2000, and C/BD-2000 (transport only) all use the same CEC transport, the CEC-Sanyo SF-P1 / SF-90.  So guess which model is the easiest path to belt drive CD nirvana?  In the Pioneers, the PD-75, PD-77, and PD-95 all use the ultimate "magnetic stable platter" PWY1004.  CDM-1's can be obtained in a number of Proceed models.  My old Sony 507ESD uses a KSS-151A.   The later Sony 990 favored by a friend sometimes used a much cheaper looking KSS-240A, which is among the most widely used transports ever.  The expensive but problematic SCD-1 mechanism is the KHS-180A.  Teac (the parent company of Esoteric) makes some models that claim to have a VRDS mechanism like the big dog Esoteric's.  But they don't, they have the VRDS CMK which looks like a barely warmed over Sony mechanism, which it appears to be, as the transport also includes the suffix KSS-151A--the same as my respectable but ancient CDP-507ESD.  Interestingly enough, a number of well known Wadia models use the VRDS CMK KSS-151A transport also.  BTW it was a broken spring, not a broken rubber band, that I replaced with a weight.  The original spring had gotten too weak for the mechanism door to work.

Here's a discussion of the various top Pioneer CD players from the 1990's with Stable Platter.  The biggest of the big dogs is PD-93, followed by PD-95, PD-S95, and PD-75 which all use the same Magnetic Stable Platter mechanism I listed above.  And also the PD-91, PD-73, and PD-77.  Other lesser Stable Platter machines include the PD-65, PD-54, PD-52.  The Burr Brown PCM63K equipped models are the PD-93, the PD-73.  Much as I enjoyed the discussion, it's critical to check back on the mechanism list to be sure what is what.

Question: Do transports even make a difference?  Well if they don't, or if even if they do and systems based on USB or Ethernet are better, we could just kiss transports goodbye, except for the "fun" of putting discs in them, and for that we could just get the cheapest available unit (not sure of how many there are anymore, but tons of old classics on eBay).  Except I personally get more satisfaction from putting my CD's in a big heavy expensive and rare player, especially if it was widely regarded as being the best.

For most of the time since CD was introduced, I haven't believed that transports make any difference, and I still don't necessarily believe they do.

As long as the bits are being decoded properly (and it is usually assumed they are, but I think periodic maintenance testing may be called for) the only difference the transport could make would be in the timing.  Rather than emitting the digital samples at a perfectly stable rate of exactly 44.1kHz, it might vary slightly.  This is called Jitter.

And this can (in principle) be measured directly or indirectly (it is usually measured indirectly by looking at the distortion spectrum), but few people do these measurements and publish them except for John Atkinson of Stereophile.

The issue that separates the objectivists from the subjectivists is not whether or not something like jitter is possible, it certainly exists!  The issue that separates the two camps is whether jitter in actual digital systems is audible.

The best and just about the only objective research on that topic was done quite awhile ago and published in JAES.  Basically they found that jitter because audible at levels about 10-100nsec, which is about 100-1000 times greater than is found in typical digital audio systems (say, 250psec).

So that's it, and I had pretty much accepted that view for the last 15 or so years, and even before that I highly doubted the audibility of jitter.

While I strongly doubt that there is this kind of difference between players, I still want the best one with the lowest jitter and the most substantial build quality and design quality, etc., just because I deserve it just as much as people who think, wrongly, that they need it.

Anyway, here's Lampizator's take on the audible differences.

I have this different theory that like looking at nice stuff, and then we enjoy the music more.  But anyway, here's another discussion on the transport audibility.  Many in this group are using a what is a pedestrian DVD player now (the Denon DVD-2900, which may have originally sold for $1000 in 2002 but now typically sells for about $100 used) as their CD transport if not their CD player, and more focusing on getting good (or at least interesting) DAC's.  I would generally agree that the DAC is more important, but that brings up another thing--according to objectivists all DAC's should sound the same also, given distortion below 0.1%, except for NOS dacs which have distortion well above 1%.

My heart is warmed by seeing one guy in that group (and I know this is fairly popular) using a Sony DVP-9000ES as transport and/or CD player.  Because that's the machine I will be getting tomorrow.  To me, however, it doesn't look special either as CD player or as transport necessarily, I am buying it for the very serious 1st generation stereo SACD capability, said to be among the best.  As a transport, it looks to use a commodity laser and similar mechanism to that in other DVD players.

To me, a CD transport should be a dedicated CD transport and go beyond the call of duty in every way possible, such as holding the disc flat and not flopping and shaking.

I have seen what most people have not, the insides of a player being adjusted, and it's not a pretty picture, and it does even lead you to wonder how the CD system can work at all.

Of course the bottom line is stability, error free reading, and lowest of the low jitter, and also low hum and noise (because even on the SPDIF output, that contributes to jitter also).  By most reasoning, it doesn't matter how much metal the transport uses with the intent of reducing jitter, it's the actual jitter performance which matters, not the intent, expense, effort, or intelligence that went in to reducing jitter.  The problem is that it's not so easy to measure jitter and errors, and it can't be done all the time until we retire the player many years later (if it's a good one, and possibly never if possible).  So we look to surrogates of good performance, like "build quality", or fancy mechanisms which clamp discs tightly or are claimed to reduce vibration in some way (which could all be marketing hype, as it could have been for the fixed laser transports).  The more effort and expense was put into making something better, the better we think it probably is.

So I laugh with Lampizator when he laughs at the real deal VRDS-NEO transports..."lots of heavy metal, if you call that engineering."  But I also laugh at him, and me, for dissing all the cheap lightweight plastic transports anyway.  Of course it should be understood that most intensive engineering actually goes into designing the plastic mechanisms.  They take lots of engineering to do correctly, including perhaps such things as finite element analysis.  With big heavy slabs of metal, you can just pile it on to your heart's and wallet's content.  And if Esoteric is doing the piling for you, it's going to cost many kilo bucks.

(Lampizator is very inconsistent, BTW.   On the one hand he liked the Theta Data transport because it's a big Laserdisc transport with big motor.  Elsewhere he says smaller motors are better for good sound.  And so on.  He's unbelievably consistent in another way.  Everything he modifies becomes the best thing ever.  But he's fun to read and especially the dissections of CD players is interesting.)

Lampizator did an interesting dissection of the Audio Alchemy DDS Pro.  According to him it is basically just a Pioneer PDS-502.  And he think compared to standard mechanism, the cheap version of Stable Platter is all plastic, every significant part is plastic, including spindle and bearing and it wobbles like hell, and though it works fine, ordinary middle of the road Sony and Philips transports of the time were high end by comparison.  This makes me feel good I got the PD-75 with the largely metal Magnetic Stable Platter which has features like a heavy metal platter and spindle, sapphire bearing, and a motor with far more torque than most players.  Below, however, Stephen Sank has a different view, he recommends the Audio Alchemy to others even though he has one of the big dogs.

BTW, regarding the Denons, the transports are generally metal.  The HOP 1000 mechanism in my DVD-9000 is all metal, looks solid and ok, but not overbuilt though, the cast metal early Philips CDM mechanisms looked far stronger (from pictures).  And there isn't necessarily anything special wrt how the CD is gripped.  I think that detail is too often overlooked, which is why I like the Pioneer stable platter idea.  Even top loading CD mechanisms that require you to hold down the disc with a magnetic weight still have essentially let the CD flap in the breeze.

Did Pioneer lose the 1990's CD wars because Sony had better sound, as The Vintage claims, or better marketing.  I think I'm more cynical.

A guy at DIYAudio did distortion spectra testing with a number of CD players and DAC's.  Guess what, one of the best of the bunch was the PD-75.   TDA 1541A looks pretty awful actually, and TDA1540 one of the very worst.  PS1 looks horrible.  1702 is not good at all, 1704 better but not perfect.  His favorite chip is a particular PCM56 he found, which adjusted looks just about perfect, but he notes that other PCM56's have been far worse.  PCM63's in PD-8500 looks awful.

Strangely however, people find the bitstream chips in the PD-75 to sound particularly undynamic, and the TDA1541A is more popular than ever.

It really looks as though people prefer distortion.  But perhaps the guy above didn't do comprehensive enough measurements.  It turns out The Audio Critic did a review of the PD-75 when it was introduced and found very clean highs but excess distortion (still at levels believed inaudible) in the bass.  That does suggest something wrong with the audio section on this unit, but not likely the transport, as if the transport were adding excess jitter that would be as much or more evident in the highs.

In that review, the reviewer said he saw no advantage in putting the disc upright (or clamping it onto a felt platter, though he didn't mention that) except that less dirt would fall on the laser.

Unfortunately it appears that some time after its introduction, while much loved, the Stable Platter transport became infamous for dropping lasers onto CD's, or falling apart during shipping.  Famous equipment repair and modification authority Stephen Sank advises strongly against PD-75/91/93/95 for that reason and one other.  He himself uses the PDR-09, which features the same fantastic mechanism but with a much better laser.  PDR-09 is even more high priced and less attainable than PD-93.  For the rest of us pedestrians, he suggests the PD-S06, which has the more plasticky PD-65 mechanism but with a very good clock, and PCM1702's, very highly regarded for sound.  He says it makes a great transport, but you'll want to use it as a player because it sounds so good.

Well guess what there's a PDR-09 for sale on ebay right now which I could have budgeted for and might get someday.  The PD-S06 is widely available on European sites according to HiFiShark but not on US eBay.

Lampizator also likes the PD-S06, though he started out with his usual complaint about the supposedly-great "stable platter" mechanism being nothing but a plastic job like any cheap player.  But he doesn't inform us this is hardly the only "stable platter" the real stable platter machines are the Magnetic Stable Platter big dogs like PD-75 and PDR-09.  If he looked at one of those he wouldn't be able to make his joke.  So here we we see even in Lampizator as with all other reviewers that we never get the full story.

Is it All in the Clock?

Rethinking the ideal transport issue, it may be that the master clock within it may be the most important concern, as both Sank and Lampizator suggest.  Whatever the mechanism does, it ultimately feeds bits faster or slower into some kind of memory buffer.  The memory buffer drives the spindle motor servo to run faster or slower to keep the buffer at some middle level.  Only if there are errors or buffer under run is the mechanism an issue (according to some).  The clock ultimately clocks everything from the buffer into the internal DAC and SPDIF output.  So it's the clock, ultimately, which counts most (or entirely, according to some).

I argue that given a particular clock mounted to a particular machine, the drive also matters as the drive speeds up and slows down, changing the rate of information being added to the buffer.  As you get close to one end of the buffer or the other, I'm concerned about some kind of quantum time compressing effect which most engineers wouldn't be concerned about.  Also there is the unavoidable fact that different amounts of current will flow in the motor, possibly upsetting the clock.

So even in my far out view, the clock is still the most important, but if the clocks are otherwise equal, then the mechanism could still have some effect.

But how good are clocks to begin with?  I have read in fact that the designers of the CD standards already knew that the clocks were going to be very important, and they set standards, with two levels of excellence.  The first is 50 ppm, and the second is 1000ppm.  Presumably any high end CD player or transport would far exceed the first standard, and would that be audible?  I would think it would be already quite good enough, particularly with an "Elite" or "ES" player, good enough to not require a clock to have jitter far less than would be audible.

At this time I haven't figured out how to convert from these "ppm" specifications to the usual "picoseconds of jitter" specification.  And then how often do we actually know the jitter specifications, most manufactures don't give them and they are highly dependent on the equipment setup.  According to many, if your biggest concern is jitter your best approach is to buy the best possible "single box" player you can get, which only has analog outputs.  Then there won't be any issue with jitter caused by the transmission of digital signals outside the box, which is one of the biggest concerns in modern audio systems, at least in part because we use (even in AES digital) single wire (or differential, which is only slightly better) signals which combine the clock AND the data and therefore lead to the possibility of the highly undesirable data-influenced jitter.

One thing, unless I can determine that my PD-75 has far less jitter than anything else I have, I may not be happy with it until it has a sufficiently upgraded clock.  I'm realizing now that my Onkyo RDV-1 has a special Apogee sourced clock and maybe I should use that player as a transport (as well as using it for DVD-Audio…for which it has PCM 1704's).

In fact, I'd been thinking is that perhaps all a person needs is a memory player with a good clock.  This sounds like it would be cheap to make, but the only one I know of is the $4000 PS Audio Memoryplayer.

If it has a good clock, and the servo is strong enough to keep the internal buffer near midpoint, then all a good transport needs to do is provide data reliably.  But how do we know it's reliable?  Sadly few players have ever provided any way of knowing.  One of these was the Cambridge Audio CD3, which had a Disc Fault light (I would assume those show the hard errors that cause fill-in, not the soft errors that are fixed by CRC, we don't really need to know the soft errors, but I think what would be best is a counter that shows both types for the current disc being played).  The CD3 looks to be a nice unit with a Philips CDM1 Mk2 mechanism.  A transport version CD3M was also made and now priced many times higher at resale.  The companion DAC unit used 4 1541A's.

Measuring Jitter

When I started this blog, I had hoped to do objective measurements more frequently than, well, the sort of rambling I had ended up usually doing.  But it seems I have things I need to say even without doing measurements.

But I'd sure love to check out the jitter of different players and/or player configurations.  NwAvGuy, the guy who did excellent audio blogging for a few years then disappeared, gives the details here.

Other Info on Stable Platter players

This report gives far more details than the short review in The Audio Critic on the PD-75.  He measures THD as 0.004% from 2-20Hz, and 0.0035 at 1kHz.  That's excellent, and not worth worrying about, however it is as much as 9dB worse than the CD format is capable of (around 0.0015), so maybe that's in line with what The Audio Critic is reporting.  However reading this review about other design features and aspects of technical performance makes it sound like the PD-95 is one of the best players ever.  I'm not so sure about that, but it's comforting to know about all the good design features and aspects of performance.

Actually the above report may simply have been cribbed from the Stereo Review review of the PD-95 published in September 1991, as shown in this collection of Pioneer reviews.

The Audio Critic review of the PD-75 complained that the distortion rose to -78dB in one channel in the mid bass, and -83dB in the other.  What are these numbers expressed in percent?

-78dB is 0.0125% distortion about 4 times more than claimed by Stereo Review (they claimed 20-20k at 0.004%)

-83dB is 0.071% distortionn, about twice what Stereo Review claimed.

BTW, I think the best you can do with CD is THD -96dB, which would be:

0.00158%

though I'm not quite sure, I've also seen -98dB, which would be:

0.00126%

In any case I believe The Audio Critic when they say that the levels of distortion produced by PD-75 would not be audible, and they gave this player the "silver" award after noting in disappointment that the PD-73 had been better and winner of previous tests.  But the reviewer preferred to buy the Sony CDP-779ES.  He had to excuse himself, because there is no reason to buy such a pricey player (it was $1900) for audible performance alone, on that score a $20 CD player would be sufficient according to his endless preaching.  But he liked how the 779ES was made, and he said that for $12,000 he'd pass, but for $1900 he could afford and wanted the undeniable quality in excess of need.  I believe this was Aczel speaking, in a later set of reviews it was David Rich who bought the successor CDP-707ES, essentially the same as the 779ES.  Or vice versa.

Adjusting EQ: Lather, Shave, Rinse, Repeat until Satisfied

I might be inclined to say I have no method when it comes to adjusting crossovers and room equalization.  When I finally have the opportunity to do it in a big way, I just jump right in.  I lose track of time, and perhaps 48 or 72 hours later, perhaps after I've reached a plateau of some kind, or more likely just run out of time before I have to go to bed to be ready for work the next day, I call it quits.  Mainly, and that's part of it, because often I keep trying to call it quits, but there's always one more problem that critically needs adjustment also or the whole enterprise was in vain.

And so audio adjustment often keeps me up later at night than intended, though that's sometimes just because of hours of reveling in good sound after a new plateau has been reached.

I sometimes write down my adjustments so I have a trail marker to get back to later.  I've often wished I had these trail markers when I didn't.  So I'd write them down more often except that it always seems that the moment I have everything fully written down I discover another area that seriously needs adjustment.  And even more often, I find in the process of writing things down that there must have been some things I had forgotten to set back to their best position, or hadn't finished setting.  So the writing everything down often takes the longest of all, and can't fully be done until everything else is finished, which is why I don't get around to it as often as I think I should.

I also hesitate to "save" settings on my Behringer DEQ 2496 equalizers.  It seems as often as otherwise when I "save" settings I end up restoring some previous settings instead, or causing some other irreversible change.  So generally I'd prefer to have settings written down before saving them.  And since I can hardly ever get to the point of writing them down without immediately finding something else that needs changing, the saving part is rarest of all.

One of the tricky things is that when you have more than one DEQ and/or DCX unit stacked together, you end up turning the adjustment knob on the wrong unit before you realize it, or pressing the wrong button, etc.  Heck you don't even need a pile of units to do that, just working one unit can lead to unintended and hard to reverse changes.

You might think after all this time (I've been playing with spectrum analyzers and equalizers since around 1979) I'd have a more systematic and conclusive method.   Perhaps precisely because I've been at it so long that I don't.  Things like Room EQ Wizard didn't exist 40 years ago.  What did exist was oscillators and 1/3 octave analog equalizers.  I didn't have a 1/3 octave EQ until much later, but sometime around 1978 a friend of mine lent me his oscillator to sweep my system.  He said it had come from James Bongiorno and was very low distortion or something.  Whatever, it was very unstable and you could only sweep slowly or it would do weird things.  But it clearly revealed what I remember to this day: 30dB peaks!  Sharp peaks and valleys even within a single octave, such as 30-60Hz.  Certainly no octave equalizer was going to be able to fix those.  And yet people had these automatic octave equalizers.  Bah!  Even something like the automatic 1/6 octave EQ function of the Behringer DEQ 2496 is not something I'd bother trying.

I did try using the Tact automatic system a few times, but I didn't like the lack of control over the process.  I would have been happy to limit the correction below 200 Hz, but it was not until many years after my unit was made that they started offering that feature.  With the electrostatic speakers there was a great impact on the highs with even the smallest movement of the measuring microphone.  When you are actually listening, you have two actual ears at different positions, and it works differently and you adjust your head position automatically.  Anyway I think it gets much less complicated at the lower frequencies where you have the longer wavelengths making small movements unimportant.  Now I have the elctrostats more less towed in toward the listener so there is less position dependence too.  Perhaps now, finally, Tact might actually work reasonably well.

Also REW.  I have done exactly one REW measurement, and I was not impressed with the results, even turning my EQ off I hardly saw any of the features in the room response I correct for.  I haven't done more REW runs but I intend to.  But I think one issue is that most automated response correcting systems deal with shallow time intervals, not the long response I see using my oscilloscope.

So, while I look forward to comparing methods, such as comparing REW with "my method" if you can call it that, what exactly is "my method" today?  Well it has even been evolving, rationalizing even as I started thinking about writing this.

Basically, it's what every method does: Lather, Save, Rinse, Repeat until Satisfied.

It's just what you choose to do the different steps.  For lather I currently use 3 things:

1) Oscillator (I like Genrad 1390, though mine needs a little adjustment, and the gap between bands around 200Hz is troublesome.  But log sweep is Essential.)  Basically you sweep up and down and hear where the high spots are so you can shave them down.  Now with bass you don't have to be exactly at the listening position, and this is convenient, since I do most of my adjusting lying next to the controls a couple feet from the listening position, though I do most of my actual sweeping at the listening position now, I have power and signal cables exactly the right length to do this (part of the new setup, I can just pull the oscillator from the rack and set it next to listening position, taking up all the slack in the cables).  It wouldn't be a bad idea to do sweeping elsewhere, but I only did listening to booms elsewhere.  More on that later in this story, as on other points I'm only touching on now.  The Genrad oscillator is far more stable and nice to sweep than James Bongiorno's old oscillator (did I throw it out??? or give it back, I hope I gave it back to my friend, because now I know it was more intended for distortion measurements than speaker sweeping.  Perhaps un-handled Sumo Charlies never got properly alighned because I tossed Bongiorno's unstable oscillator.  My role in audio history: a fluke.).

2) Pink noise testing.  This can't show you where the actual peaks are good enough for adjustment, but of course it shows the overall level which should be as close to flat as possible.  With a system like mine, if you can eliminate the room peaks--mostly peaks--you are almost flat.  Then there are the valleys...  I'll get to those later.  But the flatness I get from merely nulling peaks always amazes me.  I'm nowadays throwing generally fairly high Q notches at the response curve, and presto all the weird looking irregularities in response go away rather than getting worse.  Also, nulling peaks seems to virtually erase room decay peaks also.  Your octave EQ won't do that.

3) Music torture tests.  With some of these, if you can even play them at an intelligible level you are doing good.  In this most recent round, I've been using what came up as I was flipping through my album list first: Bass Ecstacy by Bass Erotic (actually, one of my rarely disclosed favorites).  It can get awful after listening to over and over, if not just once, but it's the best thing ever for adjusting the bass, because there's a constant bass line which is actually quite tuneful it it's not just all boom, boom, boom because of modes.

4) Real music, ultimately, of course, though I consider Bass Ecstacy as "real" as anything.

Then, once I have found an irregularity, usually an alarming peak--of which there are an alarming number in the bass response in my living room--I usually fix it by dialing in a parametric equalization function.  Both the DCX I was using and the DEQ I am allow you to dial in around 10 or 10 parametric equalization functions (PEQ) per channel, defined by their central frequency, bandwidth or Q factor, depth in dB, and direction (increase or decrease, but increase is rarely useful, and even while I am often tempted to dial in increases, even as little as I do I almost always end up cancelling them).

As I implement the crossover functions for each range with parametric EQ's, requiring two EQ's for a 24dB Linkwitz Riley (the DEQ conveniently has "HC" and "LC" responses which are 12dB/octave Butterworth responses, which can be added together to make 24dB/octave Linkwitz riley response functions) I need two PEQ's for that, leaving me 8 more PEQ's to fool around with.  8 is just barely enough and I could imagine using more.

When I first set up the new DEQ for the bass, as I recounted in the previous post, I essentially copied over the PEQ's I had previous set in the DCX.  However there was one small catch, and that was that the DCX specifies Q factor where the DEQ specifies bandwidth.  At first I did not know the conversion and guessed very wrongly, planning to reset later.  When I did find the conversion it became clear I could not translate many of my higher Q settings exactly.  And, almost unbelieveably, they cannot be set to exactly the same frequencies at 0.1Hz resolution.

I ended up resetting the bandwidths just hours later because my guesses were so wrong, I made every notch too narrow and the bass was horribly sucked out.  BTW the rule is fairly easy to remember, a 1-octave bandpass or cut filter has Q of 1.414, and successively narrower filters have proportionately higher Q's plus a bit more.  So half octave is 2.871--just a bit more than twice 1.414, third octave is 4.318, sixth octave is 8.651--just a bit more than twice third octave.  Tenth octave, which I had guessed to have a Q of about 10 actually has a Q of 14.4, which is a bit more than 10 times 1.414.  Since I've spent most of my hobby dealing with Q's in the neighborhood of 0.707, I couldn't imagine such high numbers were needed to represent the narrow bandwidths.

So neither the Q's nor the frequencies could be copied over exactly.  So that's one excuse, but mainly every time I go through these settings I can't help editing them a bit, even as to what seems more logical or reasonable or consistent with recent listening and/or measurements.  I allow myself to do a tiny bit of editing without immediate oscillator sweeping to re-test.  About 1.5 dB, or that much at a time.

The basic process of re-adjustment (not assuming a blank slate, but a previous set of fairly correct adjustments, which is virtually always what I am doing) goes like this:

1) Tune the oscillator into the very center of the most bothersome resonance.

2) If there is no PEQ in that frequency range, this is easy.  We create a new PEQ, very deep and narrow, and sweep it up and down to most complete cancel out the resonance.  Then adjust the PEQ depth back to normal volume level for this frequency range.  Then adjust the Q as to what seems right, and test by sweeping up and down to see that the center of the new PEQ isn't surrounded by new peaks or valleys.

3) If there are PEQ's in this range, a decision may be needed as to which PEQ to coopt, or to create a new one.  If an existing very close one is available, the settings are memorized and then it is made more narrow and deep to sweep as before, then adjusted like a new PEQ.  Then a re-sweeping is done to be sure this fixes the overall problem without making matters worse.  Often a compromise PEQ frequency is chosen.  These things are very strange.  PEQ's interact among themselves and resonances in whack-a-mole fashion, so you squeeze down at one point and the resonance pops up again on this side or the other where there wasn't much resonance before.  But often you can do a pretty good job just by moving the center frequency around to exactly the right compromise point and making the PEQ bandwidth wider.  Other times it may obviously better to create two seperate PEQ's, though nowadays I can't do much more of that, since I'm reaching the 10 PEQ limit of my DEQ.

When I started playing with DEQ units and PEQ's in 2005, I studiously avoided high Q PEQ's.  There's some sort of audiophile truth or myth regarding high Q filters having time smearing properties or something.  But the situation may well be in many cases that if you are cancelling out a high Q resonance, you need a high Q cut to properly deal with it, and result can be close to perfection in the time as well as frequency domains.  About 16 years ago REG of TAS (their most respectable commentator IMO) countered the audiophile myth by pointing something like this out, that (well, according to him) room resonances are typically of the minimum phase variety, and minimum phase filters can be perfectly cancelled by the proper counter-filter, and most importantly that doing so fixes the time domain as well as the frequency domain simultaneously.  And I have observed things like this--except maybe not so close to perfection, but it does generally seem to work this way.  So you are best trying to deal with these things in an engineering-like fashion, computing the needed change and applying it, rather than fiddling around the edges with audiopile tweaks.  And so, like REG, and Linkwitz, and many others, I am a believer in the systematic use of EQ to correct ultimate speaker and room response, and not a believer in the "simpler is better" type of mythology that eschews solutions like digital EQ.  (WRT the minimum phase of room modes...I didn't believe it at all when I read it, but it has turned out to be sufficiently true to make it a good starting point, even in my listening room which inconveniently-for-theory adjoins other parts of the house on two side, further complicating any major modes.)

In fact my recent opinion may be that the higher Q the filter the better, all the less for it to mess up the general frequency response curve, if it doesn't need to be messed with generally.  And quite often it seems, even using the highest resolution scanning possible, with a hand-swept oscilloscope, that cancelling out a deep resonance cancels out the hump surrounding it also, and that's preferable than trying to cancel out the hump with a wide filter, which usually doesn't work very well, only cancelling about half of the hump when you think you are cancelling it exactly.

And the reason why these high Q filters work best is simple--room modes are particularly high Q resonances!  Even when they seem to have broader impact, a room mode is fundamentally a notch resonance.

Now as many times as I've EQ'd the living room, I had really not fully come to terms with 32 Hz or what it means.  The response at 32 Hz is quite a bit recessed at the mid-front listening position.  I've sometimes tried to boost it in one channel or the other, the left channel seeming to handle the boost better.  Just a little boost simply because 32.70 Hz is C1, a very important musical fundamental no good system should miss.  But then within a week I usually end up cancelling the boost after bloat or house rattling has become a problem again, and removing recently added boosts is the first line of defense against house rattling.

I should have though more fully about what the room modes are.  The length and width of the room are about 15.5 feet and 13.5 feet.  That 15.5 foot width creates a major mode at 2Wf or about 34 Hz.  Since the woofer is at the side of the room, the mode causes cancellation in the center and maximization at the front and back walls, leaving to maximum room rattling and discomfort in background listening.

So I can't boost this frequency at all, it generates discomfort when the room is rattling, and the resonance also induces a kind of haze which smears everything, including the midrange, even when there is no rattling.

In fact, as of today, since I finally figured this out yesterday, I've gone to cutting 32.7 slightly, about 5.5dB with 1/7 octave bandwidth.  This precisely seems to remove the wall rattling that occurs in sine wave sweeping even at what seem pretty reasonable levels elsewhere.  I could use as much as 15dB cut, but 5.5dB is a working compromise which works up to my "test" sweep level, which is louder than actual bass in most music.  Usually I try to limit compromise cuts that compromise the listening position for elsewhere to 3dB, but this case is extremely important.

Making this cut did something very good in the room resonance character.  Where previously there was a consistent resonance around 32 Hz which decayed less than half the speed of decay elsewhere, like a sore thumb, now the decay is pretty much in line with everything else.  So it is actually helping the music everywhere, even if it is causing suckout of 32 Hz.  BTW I am only applying this cut in one channel, the right channel, which seems to work because that channel lines up with the important right wall which leads to the kitchen doorway and the water heater tank and HVAC coils.  I so hate it when the heater closet is rattling, I don't think it's good for the equipment in there.  I also wonder if recent wall rattling was responsible for some loose paint around the hall ceiling.

The added midrange clarity is huge.  Of course I've always known you can simulate midrange clarity with overall bass reduction.  But then you miss the bass.  Currently the only significant cut below flat response at the listening position is the narrow 32 Hz cut in the right channel.  Generally the bass is just fine, strong and especially clear now also.

In fact as compared with the electrostat-bass style alignments (flat response in the subwoofer region, no bloated "room curve") I've been using recently, the current adjustments have way more bass power impact than most of them (except the horrid boosty ones that had to be turned back within days).

Not to mention bass impact, which is the best ever.  In fact overall clarity, impact and transparency, and I can play up to real listening level now, reaching 80dB peaks in splA and 90dB peaks in splC.

Some are going to say this is way too loud, but some cars are louder on the road.  On the street outside my home you'll never measure above 40dB splA from my system, which means I meet noise codes 24/7 at max volume, 0dB, inside.

But it certainly does sound loud, and I can play much louder than before without a kind of haze likely caused by resonances kicking in.

BTW another key frequency is about 38 Hz.  That's caused by the side wall distance.  I've been nulling that for years without realizing why.  It peaks at the listening position because the subs drive it from both sides, making the center the peak, and I sit right in the room center.

I still don't understand why I have to null at 45Hz, I've been doing that since I started setting EQ's manually in 2010.  45Hz or nearby has always needed a deep and sometimes wide null, though now less wide as I also null at 38 and 52.

Modes aren't just front to back as we think of them, they're side to side, opposite corners and so on.  There's really no simple explanation, it's just the totality of reflections inside a very complex shape, with multiple rooms and walls of varying flexibility and so on.

Now at least I understand why 32Hz is weak, and trying to make it better quickly goes awry.  This is perfect justification for getting the Bag End E-Trap, the most space effective solution to sell defined modes, and I don't have any space for 4 feet of damping in all wall corners, or more than just a few, and that would still hardly absorb 32 Hz.  An active bass trapping system is essential IMO and it's a wonder they aren't more standard.  Most are still trying to tweak their way to heaven rather than engineer it.

The above shows the current uncorrelated pink noise frequency response.  I would have no technical difficulty in getting flat or even highly elevated response below about 40Hz, but I simply have to cut it off or let it be cut off because of how the 32 Hz depression results from the fundamental room mode and if I boost the response there the entire room shakes.  But still I have useful response if not flat response to 18Hz or lower, as determined with oscillator.  (Sweeping with the oscillator it's clear the mode cancellation cutoff begins just below the highly suppressed peak at 38 Hz.)  When I can fix the room mode with an active cancellation device and/or passive damping or other means, then I may have flatter response below 40Hz.

The above also shows the result of many successful equalization victories, such as attacking the 100Hz bulge.  100Hz is where the crossover frequency so I wondered about the origin of the bulge.  I was trying to suppress the bulge by  dialing back the subs, but even dialing back to -15dB at 100Hz didn't completely help.  Then I got the bright idea (and actually this should always be done) of measuring the panels and the subs separately.  Most of the bulge was actually coming from the panels, and I now think it's primarily the floor to ceiling resonance.  Since I'm sitting within 3 feet from the floor, I get much of the elevated boundary response at the listening position.  This was a rare case when I chose to apply EQ to the panels.  I do dial back 100Hz at the subs also, so the acoustic response for both seems about correct, about 6dB down from plateau at the crossover point, according to my ears.

The 220Hz depression is almost certainly caused by the primary panel reflection to the wall behind it and there's not much I can do about it.  I applied EQ to lessen a huge peak above 500Hz and that makes the depression seem much smaller now in relative terms, and the whole spectrum is beginning to look reasonably flat.  I'm not sure what causes the 500Hz peaking, it could be another effect of the primary panel reflection with the wall behind it.

In the highest frequencies, 20kHz looks right in place with no peak before it at 18kHz, but no huge peak at 20kHz either.  THAT took a lot of work too, and the super tweeters now sound much better with a kind of quasi 6th order response curve I cooked up.  It sounds far better also, with no metallic sound ever.  They often have little sound at all actually, but that is by design.  I still have the benefit of less beaming in the sound as you move around.  Super tweeters are part of my magic portfolio.

Overall there is very little boost, if any, at low frequencies as compared with the midband.  This is the "electrostatic bass" sound which sounds real to me and not bloated.  And meanwhile I couldn't make it have much less output through the bass above 40Hz without it sounding seriously thin.  The point of best sound is often surprisingly narrow.

Now the above is something people usually don't measure...the response in the adjacent room.  This is the frequency response from the kitchen table.  In some ways, you might consider it as good as or even better than the living room response.  It's very nice actually, and here with a bit of a boost at 32 Hz rather than a 6dB drop.  Of course if I attempted the cure the 6dB drop in the living room the peak in this response would be much greater, and it gets worst near the kitchen passageway and the adjacent heater closet.  Anyway, this adjacent room response is surprisingly good I think, and it matters because I most often listen to my living room system in the background...from the kitchen table.  I could play the kitchen system, which is high end too.  But when I'm on the computer at the kitchen table, I prefer my music a bit more distant, with the living room being just right.  I know, for all my talk about preserving sonic information listening to the stereo in another room is the opposite: a form of information filtering.  But indeed that's what I want mostly, I just don't want a system that filters unconditionally.

Since I can never seem to get the numbers fully written down without starting to change them, and it's a lot of work to write them down, I'm just going to take pictures of the DEQ screens, starting with the two subwoofer PEQ's, left and right, followed by the panel crossover and then the super tweeters.

 

Looking at all the notches I've added to the sub response, you might wonder, as I have at times, if I wouldn't be better simply setting the subwoofer level lower, because it seems like I'm notching out most of it down to about 16Hz.  I've tried setting the overall level lower, but you get sweeps that are just as uneven as with the overall level higher, since most of the response below 100Hz is from the subwoofer, largely due to the 100Hz LR24 crossover.  There simply is no substitute for nulling out the resonances, and you should note that the nulling isn't even at all, it's concentrated where the resonances are, unfortunately, concentrated.

My previous setup had a grevious error that had existed for many months, the left sub was out of polarity.  The polarity difference had started when I replaced the subwoofer amplifier.  Originally I compensated at the panel of the DCX by inverting polarity there.  I remember doing that.  But when I replaced the DCX crossover about 10 months ago, I may have forgotten to carry over that setting.  Or also I had been fooling around with polarity inversions, and possibly I got confused as to which way is the right way.

Sadly the DEQ's have no polarity adjustment, but I am using the digital phase control on the subwoofer at 180.  I've always distrusted these controls because I wonder what they do at intermediate positions.  In general people should be adjusting delays to get time alignments between speakers, not applying frequency-dependent "phase" adjustments.  I think when set to 180, however, it simply inverts polarity, which is what I need.

Needless to say, fixing the subwoofer polarity had a profound impact on focus, it is so much better now.  People think of subs operating below the crossover frequency where hearing has little directionality, but they also operate above the crossover frequency to some degree.

Why me, I wonder.  Why am I saddled with a living room with 33Hz suckout when I am a bass loving organ music lover and 33Hz is a key note?

It does look like the "large room" recommendation has some merit, in fact my living room is smaller than some home theater standards recommend.  It looks like walls should have a minimum length of 18 feet or so, to put the primary resonances well below 33 Hz.

Of course, it's also an opportunity for me to solve universal problems.  Most people in the world cannot have mansion style living rooms not to mention dedicated THX specification listening rooms.

Now some audio-simplicity scolds would say I need to rethink this differently.  I need a whole different arrangement (as if I hadn't thought of that).  I need no sub (no thanks!!!) or one sub placed in different locations (where?) or maybe lots of small subs with no EQ.

Yes I can see that things might be different if I put the subs in the plane of the listening position, but that would destroy the functionality of the living room as an entertaining room.  I have to work within the constraints of multi-purpose rooms, as most people do.  Again, another "opportunity" to be a leading problem solver.  I've thought long and hard about different arrangements and multiple subs and nothing other than what I'm doing works very well for my room as both a listening and an entertaining room--except for the use of a resonance canceling device or devices in the back of the room, and that's the tack I will be taking.  I've enlisted EQ to do all that EQ can do, but one thing it can't do is cancel a suckout caused by the primary room mode.

I am also perhaps unfortunately constrained by my choice of subs.  The PB13 Ultra are Big, and I have two of them in a small room.  Having such large subwoofers creates additional constraints on where they can be located without destroying the multi-functionality of the room.  Things would be different if I had much smaller subs, and perhaps something like the $2k "swarm" of 4 small woofers I that REG of TAS touted recently.

To defend this choice, I've always wanted to do real subsonics down to at least 16 Hz.  You'd need a lot of small subs for that.  I've always wanted the sub to have far more capacity than needed, so it wouldn't sound strained or be limited in dynamic range.  It's amazing how much air moving subwoofers have to do for real subsonics.  If you really want to do subsonics well, no cone driver of any size will ultimately be enough.  The best subwoofer ever made was based on an electronically controlled fan.

So this puts me in the sorry spot of reaching for 16 Hz and finding that makes it difficult for me to do the more prosaic 32 Hz also.  In fact this points out that one can (and I quickly did) reach the point where it's the room response that is not only the limiting fidelity factor, it's the limiting factor in dynamics AND fidelity.

So now I can recount the sad tale of when I first got the PB13, I couldn't make it sound good at all.  It was a one-note charlie, and just buzzed and rattled with every frequency I put through it.  But then I discovered that all these buzzes and rattles actually were coming from my room, and not the subwoofer.  And that set me down the path I'm on today, of using EQ extensively.

Actually this also brings to mind my first subwoofer purchase in 2005.  I wanted a more audiophile-recommended subwoofer, so I got a PSB subwoofer from a dealer in Austin, which required me to rent a van.  This was for my bedroom.  I set it up and got nothing but rattles and buzzes.  I rented another van and brought the two subwoofers (one never opened) back to the dealer.

I seemingly had less trouble with the PB10, a very inexpensive sub from SVS, though by that time I had figured out that most buzzing didn't come from the sub itself.  Finally I got SVS's best cylinder sub for the bedroom, and that has worked incredibly well.  By contrast the PB13's have been very problematical in the living room.  I sometimes wonder if cylinder subs would work better, but I think the main problem is the living room dimensions.

NOW there is one more trick I can try with my 2496 DEQ.  I can apply dynamic EQ, that only notches out 32Hz when it goes above some level.  I think I'm going to try that.  But I've also ordered a Bag End E-Trap now, and I think that's the main way I'm going to fix C1 on my system.

Update: Well sucking out the major mode is immediately obvious, the stereo clearly speaks with less authority in the room.  The stereo has to stand up to the plate and make the room swing just a bit.  I had to compromise again, dialing back by 3dB so there are now only 3dB and 2dB cuts.  Dynamic eq might work if set up properly, or perhaps a Levinson-esque approach of presetting the 32Hz cut for the music at hand.  For regular music just a little cut for taming the room decay, for torture bass discs a larger cut to keep the room from exploding.

Also there was a certain lack of sweetness which made me remove the two notches around 500 Hz that seemed obviously necessary from sweeping.  Will have to determine what these resonances are caused by.  Now the panels only have the 100 Hz cut (and more could arguably help) and the 4kHz intentional roll off which might actually just barely get below flat from 4-6k.  The rest on the panels is au naturale for now and the 500 Hz may be a natural thing and explanation for the damping panels in back.

That's how this goes, yes I admit even for measurement head like me listening and guestimating is the final word, until the next measurement, and so on.

Later Update:

Wow!  Dynamic EQ is having your cake and eating it too!  No need to attenuate at the primary 32Hz mode, except dynamically after 20 ms (or less, perhaps) and and after a particular threshold.  It doesn't sound unnatural, it sounds far more natural than just notching out 32Hz.  It brings back the "authoritative" sound of the stereo to at least let it excite the room mode up to some level, even if limited above that.

I was really missing 32 Hz, which indeed appears to be a very important fundamental to be missing.

Still, it would be better not to need this form of dynamic compression, so I'm looking forward to playing with the E-Trap.  What the E-Trap can't solve may be relegated to compression.

Here is a incredible Room Mode calculator, amroc.  To get numbers similar to what I actually measure, I have to enter length of 17, width of 13, and height of 9.  The actual numbers are 16.5, 14.5 and 8.5 (average).  The actual numbers are probably affected by the hall on one side and the kitchen passage on the other boosting the effective length, and the width being reduced by bookcases and furnishings. Still it clearly shows the complexity and a very similar resonance spectrum to what I am dealing with in any case. It's not all in my imagination.

Nyal Mellor, a major regular at What's Best Forum, has written a good short intro to room acoustics and room correction here.  This is straight ahead, no BS.

Art Noxon, seller of acoustical damping treatments, makes his point about the limitations of EQ, and as other such sellers sometimes do, goes a little OT.  EQ doesn't change the fundamental resonances and their impact on the sound...well actually it does change their relative impact on the playback sound, by making the resonances far less objectionable to musical playback, and to a degree working "with" the resonances rather than ignoring them.  And also, there is such a thing as a sound cancelling speaker, and I'm beginning to think something like this is key to playback in small rooms.

It's possible the most excellent cancelling would be done from a system that operated from the playback source.  Swarm type subs do this.

Come to think of it, cancelling the primary mode is simple in these terms: put a 0 degree phase woofer in the antiphase of the resonance!

I see issues however.  First I'd want this to be a specially restricted woofer, that only speaks to the resonances, and doesn't cross over at the front sub crossover frequency of 100Hz because that would introduce too much imagine blurring spurious out-of-plase high frequency sound.  It could just cross over much lower, or be tuned.

Being tuned in this way, it doesn't have to "sense" sound through a microphone.  (And, it also has the advantage of more time to "think.")  But other than the increased time, the same FFT type convolutions need to be done, either on a "sensed" signal or an electronic one.  If one is just reading the signal as it comes, it doesn't matter fundamentally whether the resonance cancelling satellite sub is reading a microphone or reading the stereo audio signal.  If the microphone is reasonable accurate.  And what is sensed is what actually is, rather than what one predicts one needs to act upon, which is actually too impossible to calculate.

So I point this out so as to make it clear that a speaker with the sound signal isn't necessarily totally better off than one that sense sound in a modal corner, for example.  Though, in the long run, subwoofer satellite resonance cancelling speakers could take advantage of both.

I do regret the rather high price...but I'm being a pathmaker here.  Most people are way behind in discovering the need for things like this.  If there was a mass market for modal cancellers there would be less expensive options.  Though I'm thinking the Bag End company has good experience for making devices like this.  They famously made an "Infrasub" with response to 8 Hz.  Such a speaker would be impossible with ordinary cones (electronic fans could do it) except with extremely good limiting.  So the speaker has 8Hz response most of the time, giving you the ambiance, but not the full 8Hz special effects of a real life explosion, for example.  That kind of thing has it's place, and in fact that's another thing that it looks like all good systems need: dynamic EQ to keep the system and room from exploding.  So you get the full everything up only until you can't.

And then the nagging voice about the money, yes the money.  I have a clear need for 32Hz response, as a bass lover, organ music lover, rock music lover, symphonic music lover.  Circumstances have cheated me of that response, but there's a sensible workable fix, the Bag End E-Trap, it just costs $1650 (incl shipping).  Others might spend a fortune on tweaks to fix the problem and get little or no real results.  I'm headed right at the heart of the problem and taking an unusual but apparently workable solution, and there are no other workable solutions that wouldn't mess up my life.  This is an apparently necessary companion to my pair of $1750 subs.

Setting up the new Bass crossover

Stack of DEQ units on right is new Crossover

On Labor Day weekend I finished the last part of the new living room system crossover design I have been working on for the last couple years.  I removed the old Behringer DCX 2496 digital crossover which has only analog outputs and installed the new Behringer DEQ 2496 for the bass section.  The DEQ has digital inputs and AES digital outputs which are connected to a new Emotiva DC-1 DAC which drives the two subwoofers through balanced cables.  So now my 3 way system uses all Behringer DEQ 2496 digital processing and connections for the crossover section and 3 high quality specially chosen DAC's for each way.  The Emotiva DAC is the only Sigma Delta type DAC in my system because I feel that Sigma Delta--which I believe corrupts high frequency timing integrity and thereby loses or fakes high frequency information--is appropriate for and only for the subwoofer section.  I was hopeful that the DC-1 might be direct coupled, but I see that direct coupling become unpopular recently (Oppo went from direct coupled to capacitor coupled in their latest TOTL player, the BDP 105 because of compatibility issues, so I suspect Emotiva has followed similar reasoning) and Emotiva consistently never says whether it is direct coupled so it probably isn't.  Anyway the Emotiva is small and very high quality (very good, though not the very best, noise and distortion specs--which are given quite completely) and had key features I need (AES input and balanced output) at amazingly low price for an audiophile grade DAC and the digital volume controls have been useful.  I determined in a previous experiment (not yet described here) that the volume control on the DAC has less noise than the DAC itself because lowering the volume equally lowers the (very low, but still detectable) output noise level for the first 20dB or so, and the noise is already lower than my Onkyo DAC at 0dB (but not as low as the inputs of my Parasound power amplifier or my Meguro noise meter--making it easy for me to test and make this finding).  The Emotiva has balanced AES inputs and balanced audio outputs which are useful for the subwoofers.  I like it a lot but I could have gotten a truly direct coupled DAC of similar quality from Ross Martin audio for slightly cheaper but the Ross Martin version would not have had the useful digital volume controls which I figured I was going to need for now.

New Emotiva DC-1 DAC on top for Subwoofers has all balanced connections, Onkyo 1704 DAC below for supertweeters

Actually the crossover change was slightly complicated by the decision to put the newest DEQ on the top.  The newest DEQ had a slight chassis warp which I fixed partly by readjusting the rack handles and partly by setting on only 3 feet.  Each DEQ unit sits on rubber feet.  I stick together three 1/4 inch high feet very much like the old Radio Shack feet (which Radio Shack no longer sells, but Newark has an almost exact replacement) to make each foot for each unit and I normally use 4 feet .  But on the top DEQ unit I use two feet in front and only one in back, which eliminates any wobbling.  Well clearly this unit sitting on only 3 feet needed to go on top so as not to endanger the others.  And I figured I wanted the unit on top to show the tweeter response simply for intuitivity.  (BTW, unless you jiggle the stack of DEQ's very hard, you wouldn't realize it was anything other than perfect, it looks perfect.  And that perfect look took a little extra work on my part that I'm describing here.  When I started, the warped unit made the whole stack look bad.)  And since I decided the top DEQ should also be the supertweeter unit, I had to copy the supertweeter adjustments from the #2 DEQ into the newest one before copying the subwoofer adjustments from the DCX being removed to the new #3 DEQ.

For the AES cable to the new DAC, I used a 12 foot Mogami Gold which was plainer looking but actually more expensive than the Geistnote cable (which is made using slightly less expensive Mogami balanced AES wire) I am using with the Lavry AD10 ADC.  Actually I would have gotten another Geistnote cable but they don't make one in 12 foot length, only 10 and 15.

After setting everything up, the real fun begins.  I have to set up all the parametric EQ's, and I use parametric equalization extensively in the subwoofer bass, in addition to the crossover function itself which is realized using parametric EQ's.

It's only when you get to doing this that you realize some limitations.  The Behringer DEQ units are highly flexible, but their only gain control is stereo-only (there is only one gain control, in the Utilities Menu and it only works in 0.5dB steps...I can live with that because I don't think my adjustments are ever more accurate than that anyway, but 0.1dB would be best).

Fortunately, the SVS PB13 Ultra subs have their own gain controls.  I primarily use the left one which has the replacement amplifier with a digital gain control.  I like digital gain controls because the settings are reproducible.  With analog controls, if you accidentally turn the control (it happens to me all the time) the adjustment is lost, and you now may have to go through a complicated "adjustment" process to restore it to the previous setting and it may take hours until you are happy it is done correctly.  So I don't use the gain controls on the back of my supertweeter powering Parasound HCA-1000A amplifiers--I have found that to be especially dangerous because reaching behind the amplifier you might think the volume controls are the speaker binding posts.  In that case, where the gain is much too high for my intended use...and results in excess noise audible with ear to speaker...I use 6dB gain reducing RCA adapter plugs.  Now 6dB isn't quite enough, I could use as much as 12dB though 9dB might be optimal.  I have 12dB reducing plugs, which I had been using in the application, but I ultimately found they added distortion.  I'd love to make my own 9dB reducing plugs someday, or maybe just mod the amplifier.  With this 125W rated amplifier, full power is reached with 1.1V input.  By comparison, my 300W rated Krell requires 2.25V input for full power.  I feel this is too much gain in a lot of cases.  In mine especially since I don't need more than 25W for the supertweeters.  John Curl told me his hands were tied because of THX requirements.  A real nice amp would have control with discrete steps, which could then switch in a potentiometer as one of the steps.

I keep the analog volume control on the right subwoofer in the middle, pointing straight up, because that is a fairly easy position to reset.  The corresponding level on the other side is around -9dB.  It might be better to chose a different level on the right and "block" the control so it naturally reaches that level when turned up.  I'm thinking -3dB or -6dB might be better.  But it probably matters little whether I attenuate at the subwoofer itself or in the Emotiva DAC.

Two New Racks

My living room has undergone a large rearrangement since June.  A new 2 seat armless sofa was added to the back of the room, with Queen Anne chair moved to the side.  The 3 seat convertable sofa was moved toward the back, where the keyboard table used to be.  Two book cases were moved out of the living room and into another room.  All this makes for better movie viewing for most guests at my monthly discussion and movie party.  It makes for better seating and discussion too.

A key part of the plan was to rearrange the living room audio system also.  The piles of equipment behind each speaker were removed and placed in two new racks on the right side of the room.  The midrange amplifier, DSP units, midrange DAC, and power conditioner are still in the center of the room, but no longer in piles behind each speaker.  I feared the piles of equipment were interfering with the sound, and my fears have been confirmed.  It generally sounds much better now.

Since late last year I had planned to get a dual width Mapleshade rack.  Sadly the time for that never arrived, and vast money was spent on other things.  I feared the setup difficulty too.  In the "meantime" I first got one brand new 4 shelf Sanus rack because I needed "something" fast to finish the living room rearrangement which had to be completed before my party at the end of July.  The $129 price is unbelievable for such a well designed, useful, and reasonably well made rack (and from an actual Sanus online dealer).  And nice looking too, possibly an average guest couldn't tell the difference between this and a serious $5000 audiophile rack.

The first new Sanus rack was set up before my party in July.  I had the rest of the equipment piled on the 3 remaining shelves of my old (and strongly disliked because of pole spacing) Sanus Eurorack.  I was finding that having two Sonos boxes and the transformer on top of my Kenwood L-1000T was just too much, that tuner gets very hot even with nothing on top.  So I decided to get a second $129 rack to at least fix that problem, and several others, as well as be much nicer looking.

August I set up the second rack.  I also finally hooked up the equipment on the first rack, notably the two silver disc players and the Lavry AD10, with all new cables, and added new speaker cables (Canare Star Quad) for the Acoustats also.  But on the first hookup weekend I found I didn't have any suitable speaker cables for the super tweeters.  So that had to be ordered (more Canare from Blue Jeans Cable) and was ready to be hooked up the next weekend, which it was.

In the upgrade phase I have just completed the super tweeter driving chain has been greatly upgraded.  The Behringer DCX digital crossover is no longer used for the tweeters, I use a separate Behringer DEQ 2496 with parametric filters programmed to be the crossover.  The DEQ gives me a balanced digital output, which I convert to coax with a box I bought from Markertek.  The coax digital feeds an Onkyo RDV-1 DVD-Audio player which has digital inputs so it works just like a DAC and uses the fabulous PCM 1704 converters.  The 2.2V analog output (not the 10V output of the DCX) then feeds the perfectly fine and unchanged Parasound HCA-1000A power amp.  Which now feeds Canare speaker extension cables with locking bananas.   Now the supertweeter has its own digital crossover and PCM 1704 based DAC and uses low inductance cables.

(In the next phase, the subwoofer driving chain will be similarly upgraded.)

The cables everywhere are far better arranged than before.  The front of the living room used to have a murky pile of cables about a foot deep.  I couldn't remove cables anymore, I simply had to add new cables, so the pile had grown.  Now I have neatly arranged and a surprisingly smaller number of cables (despite more connections actually being made).  The AC power in particular is better set up.  I use currently only use half of the 8 outlets of my Panamax MB 1500 power conditioning UPS.  Each outlet feeds some kind of strip.  The racks at the far end have hospital grade $129 6 outlet strips from Wiremold.  Currently two strips with a third now on order (after I used up all 12 outlets and still needing more...).  There's a different brand with wide spaced outlets for the TV and various power adapters.  And there's the very very nice 6 outlet power box from Cullen Circuits powering my Tact digital preamp and Behringer DSP's.

Unlike before, when there power strips plugged into other strips with intervening extension cords plugged into the power conditioner, now there is only one level of strips plugged into the power conditioner, period, and no extensions for anything.  I believe this is essential for safety (some might argue all the outlets of the condition should be direct connections to equipment, but that's ridiculously impossible, I'd need 4 power conditioners to provide that many outlets, and it's actually best to have just one) and it makes for good organization.  It helps that Wiremold makes the hospital grade strips with 15' cords.

Better, better better!  It's taken a couple months of work and ordering cables.

Analog Resampling

Don't throw away those silver discs just yet!  I know this is nuts, the typical audiophile nuttiness.  But it's my nuttiness, not your NOS DAC's and that sort of thing (which I denounce).

I continue to find that listening to CD's has a special magic that listening to digital files doesn't have.  And the system in which I demonstrate this takes the analog output of a CD player then uses a high end AtoD converter to produce digital for digital crossover, then back to analog through high end DtoA converters.  I continue to believe that taking the analog outputs of the CD player is crucial for preserving the music.  Piping the digital output of the players directly into the digital chain has a comparatively flat and sterile   sound, even slightly harsh at times.  Resampling the analog always sounds smooth, but also like the music has been preserved in some special way.  It has always sounded this way to me (though, still thinking this to be a bit crazy, I haven't bothered with any ABX testing).

Now that I have my two new racks set up, I have my two super special disc players hooked up to the Lavry AD10 converter also.  The current lineup has two classic players, the Denon DVD-5900 and the all time heavyweight Denon DVD-9000.  I use the 5900 for SACD's and DVD-Audio, and the 9000 for CD's.  Actually the official purpose of the DVD-9000 is for HDCD's, which must be decoded by a HDCD player to extract the decoded dynamic range and filtering, and the DVD-9000 is the best HDCD playback I have ever heard and may well be better than the more famous HDCD statement players by California Audio Labs which only used PCM-1702.  But, as I have just verified, the 9000 rocks extra hard with regular CD's too.  With regular CD's, the 5900 sounds merely OK.

It was using the 5900 in 2010 when I first made the discovery about how special analog outputs sound.  Of course it's also true that Denon uses special processing known as AL24 to give 24 bit character to all recordings.  Maybe it's just that.  But now I'm finding the 9000 to be far better than my well used 5900 for CD's.  It's too bad my 9000 won't play SACD's or even DVD-Audio's, in the latter case because the motor drive can't handle higher speeds anymore (I tried to have it repaired, but new motor didn't help, it needs some on-board refurbishing which factory repair centers don't do).

Even though it's actually my 9000 which is the more dysfunctional player, it has the Magic, and the 5900 might well be replaced by something newer, such as an Oppo BDP-105, or something older but specifically designed for SACD, such as a DVD-9000ES or SCD-1.  However right now the 5900 is the most magical player I have for SACD and DVD-Audio and it does still do it for them.

I'd been listening to the SACD version of Brothers in Arms on the 5900.  I needed some of that hard rocking again, digital files were sounding too ethereal.  Playing discs on a silver disc player and taking the analog output restores the palpability.  Then I switched to a plain CD of Pretzel Logic.  It sounded only normal on the 5900, not unlike the digital files, but magical on the 9000.

I have an explanation that objectivists aren't going to like.  Ignoring the dual differential PCM 1704's in the 9000, the dual power supplies, and other special features, the special reason why the DVD-9000 sounds special is because it has to.  It has no other reason for existing anymore other than to sound special.  The player is barely working, can't handle DVD's anymore, so this is its Swan Song.  Swan Songs are special.  It's giving its all, and it has a lot to give.

Anyway, I've thought a lot of why analog resampling might actually make things sound better.  It's obvious that it might be adding just a bit of noise, but that's not what I think is doing it.  I think expressing something in the analog domain is special somehow.  And it also gives me the ability to resample in a better digital format, 24 bit 96 kHz.  This is like opening up more space or something.

I think there's something like "grounding" going on.  By expressing the music in analog form before resampling, it's turned into something real and not something imaginary.  And that's kind of how it sounds.

Others have different nuttiness.  Mark Levinson, for example, has been recently selling his "Master Class" software for EQ which features something called A+.  Whatever A+ actually is, isn't revealed.

https://www.danielhertz.com/master-class-software

Now clearly this is something entirely different from analog resampling as it's an all digital process of some kind, implemented in software.  It might be something like Denon's AL24 processing--which in my experience has been good sounding.

Or not.  One interesting idea I've thought a lot about is that each time I play a disc, my resampling process creates an entirely different set of numbers.  This is because the sampling clock is running independently of the CD player clock, so it starts at a random point each time.  Plus, since the two clocks aren't connected they may slightly drift apart.

So each time I get a unique set of numbers describing the same sounds.  According to conventional audio engineering, if the resampling is done well enough (and it is) they should be indistinguishable anyway.

But I still think it's special somehow.  And, actually an entirely software-driven process could do exactly the same thing, by doing the virtual equivalent of analog resampling.