Finally getting around to the Midrange EQ

I recall now that measurements over the past 12 years (since I've had a serious living room audio system featuring Acoustat electrostatic speakers, for the the 1+1's and now since 2019 the 2+2's) have fairly consistently showed a dip (or the start of the general High Frequency rolloff) right around 1kHz.  I had often thought to myself "that's not good" but also, almost always, "I'll do something about that later when I have time for serious listening.  Midrange EQ requires serious listening, not mere measurements.  Just tossing in some EQ by measurement is as likely to make things bad as good.  Especially starting from a well regarded speaker like the Acoustats, which pretty well determines the frequency response at 1kHz since room nodes are not important there."

Starting Response, Uncorrelated

Starting Response, Correlated

Notice that of all places, there is a slight depression right at 1kHz, where we are the most sensitive!

 Now, in principle I could crank up the tweeter control on the Acoustat...but then the part above 1kHz would get raised even more, where it begins to sound edgy.  So you have to pick a spot like this in the HF attenuator, a compromise.  (I haven't actually technically explored the HF attenuator control much, I'm afraid of being able to re-create the current setting without an objective test, which I haven't yet verified.  It was chosen with great care in listening as exactly what I describe, a compromise.  Which happens to be, for the 2+2's, right at the center of the control range, as you might expect it to be.)

Most of my parametric EQualization's (PEQ's) are simply notches for notching out room modes.  The technique is not unlike what a computer would do.  I sweep up and down with an oscillator to find the worst node resonance, try to cancel it out exactly with a PEQ notch by setting the Center Frequency and Q, and move on to find the next remaining worst node resonance, until everything is generally smoothed out, and/or I've run out of my 10 available PEQ's.

I have generally avoided EQ in the 125-17000 Hz "midrange" signal sent to my Acoustats.  It has a few resonances (one around 515 Hz for example) that I've sometimes notched out, but then later undone the notches because I felt that perhaps it was obscuring the midrange, which always has sounded slightly distant and opaque in the first place, and I worried (but didn't actually prove) that the notches might be making that worse (perhaps they weren't I'm thinking now).  But I never felt strongly enough about that to actually do the serious listening to adjust the midrange EQ around 1kHz by ear.  The "raw" Acoustats are good enough, I sometimes opined.  I don't want to mess with something I'm more likely to make worse than better, and where it's important.

Another issue is that in the conventional wisdom, you should use EQ to make cuts only.  You should not try to use EQ to boost depressions, and especially not notches (where EQ generally doesn't help at all and such high levels of EQ cause other problems).  Well, actually, lots of people use a little EQ boost here or there, and sometimes a lot of EQ boost.  But I wasn't willing to try without some kind of serious listening experiment, which I was never much inspired to do, basically because things generally sound so very good (especially compared to all the various systems I've had before AND nearly everything I hear elsewhere with very few exceptions.)

Years ago I purchased a separate EQ unit to playing with midrange equalization from the listening position, a Monoprice 615031, a 31 band analog Graphic EQ featuring both balanced and unbalanced inputs and excellently low noise and distortion.  Previously and about up to then I had a odd collection analog graphic and parametric equalizers for EQ experiments I intended to do but never got around to.  Back in the 1980's I had a Technics 31 band (I actually played with that briefly, but found I couldn't change a certain characteristing "dryness" which had been on my mind then, so I prematurely sold the Technics then over the years acquired a bunch of funky not-quite-replacements, several ADC's, and an previously overused Soundcraftsmen which was always noisy and useless but I hoped to someday refurb.  I think I still might have had an ADC when I decided to get the Monoprice to replace it.

The Monoprice 615031 is simply in a higher shelf of performance than all the others I've mentioned.  Integrated circuits nowadays are nearly as good as the best possible discrete circuits such as might be in instrumentation or preamps designed by John Curl.  Integrated circuits have gotten so good since around 2004 they blow away discrete transistor circuits in almost all other gear, especially older gear, nowadays.  Especially in stuff made by Emotiva, for example, which tends to use the LM4562, one of the best analog chips (other goodies are OPA211 and AD797, with the latter being somewhat hard to stabilize).  And of course the likes of Mark Levinson, who fills their gear with the latest and greatest, though it's funny how even with all the incredibly cool parts and circuit board materials and all their specs don't even match Emotiva until you get to the very top and stratospherically priced units.  I don't trust Monoprice as much as Emotiva (especially after one of my Monoprice HDMI extender units failed in 2 years, though that has not been unusual for most makers of HDMI extenders, don't get me started on that.)

But this fine graphic EQ simply sat in storage until I had my first attempt at listening to Purple Rain by Prince, an album I bought a few years ago but had not played yet.  I was looking at my record collection trying to decide on what to play, and Purple Rain was an obvious choice simply because I had it but had not yet bothered to listen to it (many new albums sit in the collection for years before being played the first time...the album buying process has often seemed almost entirely disconnected from the album playing process).

Immediately the midrange equalization came back to mind because, after having removed my upper midrange Gundry-Linkwitz inspired depression a couple weeks previously without much apparent difference, I now needed to dial it back in just to make this album listenable.  Without that EQ depression, which was made up of 3 different PEQ's actually (because I added one on top of the earlier one(s) to make more extended and flatter in the periphery) the album was so harsh sounding it was pure torture to listen to. 

So I dialed back in the Gundry-Linkwitz inspired depression (I have been lowering the 2-10khz range) and Purple Rain was far more tolerable, but it was still very obviously sounding thin and edgy.  I thought to myself, "Prince mixed this on high end equipment, he would have mixed it just right to NOT sound thin and edgy."  This is the "test case" I need, I said to myself, to see if I can set the midrange EQ to make it sound better.

This album is a great test case precisely because it is NOT a "audio purist" kind of record, say recorded in real time to a high resolution or DSD recorder using a single stereo ribbon microphone, etc.  Something like that will sound "good" on nearly any system.  Instead Purple Rain is a very complex mixture of recordings combined on very low noise and distortion equipment mixed and eq'd to a very precise endpoint.  If that endpoint is high end enough to be revealing, but not EQ'd just right, it will sound terrible.

As I often argue, inverting much audiophile conventional wisdom, you need the very best fidelity for reproducing the lowest fidelity.  The lowest fidelity is the most challenging to reproduce to sound even good, unless reproduced by such comparably low fidelity it doesn't even matter.

In this case, I'm not saying the Purple Rain is low fidelity, it's just ridiculously overdubbed and complex, and that makes it very revealing of tiny issues in reproduction equipment of high resolution generally.

(BTW I started from the LP I bought years ago, and migrated to the Qobuz high resolution version I like somewhat better.  I need to do more LP adjustments too.)

So, on Saturday May 14 I hooked up the Monoprice graphic equalizer.  Actually by that time I was thinking other ideas I'd long had in mind.  Such as that perhaps a deliberate peak at 1kHz might add more "acoustic" sound to the reproduction, curiously the LS3/5A's used woofers with a huge 1kHz peak (which is also common in other drivers) but they did equalize it out mostly.

So I started by dialing in a 1kHz peak.  It sounded horrible at +15dB, but became unnoticeable around +3dB.  Listening at +2dB it clearly sounded better...less electronic and edgy.  It's funny how raising the midrange seems to clear up the high end more than the midrange itself.  And improves the bass too.

I then took a look at measurements (which at that time I'd basically forgotten there was a 1kHz depression).  And strangely enough, my dialed in adjustment to make it sound better coincided with flatter response, essentially filling in the depression.

So I tried more EQ's to fill it in even better.  But changes in pink noise response that initially looked better in uncorrelated pink noise than uncorrelated, and didn't necessarily sound better.

After a late night of fidding, I ended up with a pretty good sounding adjustment that was fairly small and simple and didn't make things worse in either correlated or uncorrelated pink noise.

+2dB at 1kHz

+1dB in the 1.25 and 1.6kHz bands

I dialed an approximation of that to the Behringer PEQ for the Acoustats (which does the Gundry-Linkwitz inspired dip) and it sounded just as good done that way as done in the separate Monoprice EQ, and when I added the Monoprice EQ on top of it (doing the EQ twice over, making it +4dB at 1kHz) it sounded worse.

Note here that I have no idea why these adjustments are needed.  I have no model, so I'm not following my "Adjust to the model, not the measurements," rule.  But adjustments like those can be made if listening does, and continues, to justify them.

Now I need to do a serious listening test and reconstruction of my Gundry-Linkwitz inspired dip, which was mainly configured ad hoc in response to pink noise measurements over time.

Present results suggest that perhaps you should use small boosts, if necessary for flatter response, at least below 2kHz.  Above 2kHz some sort of depression or roll-off seems desirable in the response measured omnidirectionally.

Final Response, Uncorrelated

Final Response, Correlated

Update: Re-doing the EQ on Sunday May 15

Somehow the new midrange was leaving me unsatisfied listening to KPAC on FM radio the next morning.  Rather than smoother midrange, I was feeling that there was a lumpier midrange with extra degrees of freedom.  It also bugged me that the final "simplified" graphic eq adjustment had created two peaks, one at 1kHz and one 1400 kHz.  Meanwhile there was a +1dB adjustment at 1400khz.  That was probably wrong.

Though possibly convenient sometimes, I just don't like graphic equalization.  It seems as though you are adding and subtracting resonance at frequencies that are arbitrary compared to the real phenomena you are trying to correct.  Plus rather than simply adding or subtracting resonance in the width required, all the resonances are exactly the same, 1/3 octave at present.  That just doesn't seem right to me.

I like parametric EQ, where I can set the frequency of the resonance or anti-resonance precisely and set the Q or bandwidth as well.

But first I thought I'd use my Kron-Hite oscillator to sweep the area to see what the real phenomenon I am trying to correct look like.  Nothing can give you a better feel, I have believed, than slowly sweeping with an oscillator.  I used no EQ for these tests.*

But strangely, right here at the center of the midrange, it wasn't much use.  What I was hearing sweeping from 100-1000 Hz and 1000-10000 Hz (sadly the Kron-Hite doesn't provide any easy way of sweeping around 1000 Hz because of its choice of frequency ranges, and in the regards, General Radio oscillators are often nicer with their 200-2000 Hz range, though what would be best of all would be some way of controlling the ranges themselves) was one tight little resonance after another, just going around and around and around from peak to trough every 3% of frequency change or so, but rarely changing in amplitude very much.  Ignoring the tiny peaks and troughs, which are probably caused by room reflections, the general response was very very smooth, it was hard to hear any changes at all.  Or in other words, the tiny peaks and troughs seemed to be nearly all the interesting stuff that was going on.  But I did seem to hear that a particular resonance just above 1kHz, 1015 Hz to be exact (measured with my Fluke 8060) was slightly less resonant than the resonances around it.

So I decided to start there (actually 1013 Hz was the closest available frequency I could select) with a Parametric EQ having bandwidth of 1/3 octave and +2dB boost.  It seemed to do a good job fixing the notch around 1kHz without creating a new peak above it.  But this seemed to highlight another trough below 1kHz, somewhere around 850 Hz.

I possibly didn't try hard enough to eliminate that lower trough with a wider bandwidth for the 1013 Hz PEQ but I still don't think it would have worked right.  Instead I added a second PEQ, this time starting around 830 Hz and moving it up until the response looked right.  I settled on 877 Hz, with bandwidth of 1/6 octave and height of 1dB.  Ultimately I widened the bandwidth of this PEQ to 1/3 octave as well.

Correlated PN, 1013Hz at +2dB

This looked far better I thought than the Graphic EQ adjustment.  And replaying Purple Rain again, I thought it sounded way better, though I was goaded to raise the PEQ at 1013 Hz a bit more to +3dB, and that made this album solid as well as magical (and far away from the unequalized sound which had been harsh, thin, and barely tolerable).  With Graphic EQ I had never been able to get both 1Khz and the band below it flattened without causing the periphery to rise up like mountains.  The problem is that the required correction frequencies do not like up with what's available on the EQ (and perhaps the bandwidth as well, but this time I ended up using 1/3 octave bandwidths).  (Note the periphery does still rise up a bit, but no more than it does without any EQ.)

* Method Notes

I do things in ways that would probably disturb most audiophiles.  But with purpose...generally attempting to do correct level matched comparisons (though, technically I skipped that a bit during these tests...but it would have no effect on the measured responses).

When I used the Monoprice Graphic EQ for rough adjustment test purposes, I had the Monoprice directly feeding my rarely used "A" amplifier, the Aragon, and I was using my ABX Switch to select between this equalized signal and the unequalized version fed through my Hafler 9300.  Both amps were getting their signal from the same Emotiva Stealth DC-1 dac, but I was feeding its unbalanced output directly to the Hafler, and the balanced output to the Monoprice and from that to the Aragon.

The level of the Monoprice/Aragon signal was then matched to the Hafler by adjusting the EQ level controls of the Monoprice (which infuriatingly are only engaged when the EQ is active) and switching back and forth until they sounded the same not only in level but in quality.  I did this precise level matching before adjusting any of the equalizer band controls--they were all set to flat.

Audiophiles might look at the fact that I was using two different amplifiers, not to mention all the different cables, and the equalizer circuitry, etc.  But the truth is that good amplifiers, when level matched, sound exactly the same.  And quality line circuitry as well as decent cables are perfectly neutral as well.  This has been proven over and over again by endless tests by objectivist audiophiles, a long running $1000 contest, etc.  And by my own tests with Hafler, Krell, and Aragon amplifiers--when working properly and level matched they all sound the same!  Subjectivist audiophiles don't believe it from their own experience because they almost never do proper level matching, nor do they ever do instantaneous A/B switching made possible with an ABX Switch and so have to rely on their highly inventive audio "memories" (which are really more like stories you tell yourself).

This time I didn't even bother to do voltage measurements, I simply level matched the different amplifiers by ear.  After the first pass, the level sounded the same, but the one amplifier or the other sounded different in some way, such as MORE highs or MORE bass, etc.  When one side has qualitatively more of any one thing or another, its level is turned down very slightly or the other side is turned up.  Eventually you get to the point where indeed the amplifiers sound absolutely identical.  Then you can add in a dollop of EQ, and the difference that makes is fairly obvious.

Now the place were I did not do level matching is this.  When I turned up the EQ by +2dB at 1013 Hz, for example, I did not attempt to level match THAT overall to the unequalized condition.  To Be Perfectly Fair, in evaluating which sounded better, I'd have to do that, say using C weighted full spectrum pink noise and trying to match the levels by ear or C weighted SPL measurement.  One problem here is that since the frequency contours are different, there never might never be a place where they would sound qualitatively the same, I could at best only match the "rough" level, which would probably not be much different than no matching at all (since the levels are the same everywhere except the 1/3 octave around 1013 Hz).

Since I know what my alterations are doing, I think this is still pretty fair.  Where this sort of open ended unmatched testing would be a problem is when you don't know what physical effect your change is making, and you are making lots and lots of changes like that.  The tendency will be to prefer changes that slightly increase the level, so after many such changes the resulting levels will be significantly higher than before at the same volume level setting, but then you might also be setting the volume control down over time as well to compensate, with the possibility that all your changes have done nothing, or nothing but change the level.

I also hooked in the oscillator through the Aragon for convenience.  I put the oscillator on top of the Monoprice EQ, and hooked the cable going to the Aragon to the Oscillator output.  That was a fine way to do sweeping (of only one channel btw) but then I could not at all tell the effects of different PEQ's in the Behringer DEQ 2496 (which feeds the Emotiva DAC) on the sweeping signal.  I had to evaluate PEQ changes by measuring pink noise, and by ear.

Update May 16, taking a look at that "Linkwitz Gundry Dip"

Now, what has also been on my mind for a long long time.  I've fallen into the habit of making a depression in the 2-12kHz region because, well, various reasons.

Linkwitz and Gundry recommended depressions above 2kHz

Linkwitz rationalized that domestic rooms have a lot more high frequency reflectivity than large auditoriums.  High frequencies can enter the ear not just from the front but from the side where there is a more direct path to the eardrum.  Resulting in a sensation of excess highs.

I seem to recall these dips being recommended as 2-4kHz or perhaps 2-6kHz but not roughly 2-12kHz as I am doing.

But I was also faced with some unevenness in the highs that was hard to iron out, most infuriatingly right at 6kHz and even 12kHz which are characteristic "metallic resonance" frequencies.

Anyway from rather ad hoc measurements and tests I ended up hacking away at the 2-12kHz response rather significantly.  Perhaps too much.

Some of that hacking may have been inspired by the relative lack of midrange around 1000 Hz, which I have just now fixed.  So the EQ in the 2-12kHz region now needs to be entirely reconsidered, perhaps even from scratch.

But first I needed to re-measure the system response since I had not done so after adding an extra dB to the 1013 Hz boost, making it 3dB, which really seemed to bring things alive (without being too much).

Well, now I believed I should up the 873 Hz to 3dB as well, as now a trough appeared at "900 Hz."  That didn't seem to erase the trough completely.  I believe I need to actually find the null point I am trying to correct, since I see now that the 1/6 octave depression was at 900 Hz and not 800 Hz as I had believed.

I did some sweeping and found the actual low null nearer to 853 so moved the boost point there.

Meanwhile I did zero out all three of of the high frequency EQ cuts, and I am casually listening.  It does sound a bit on the bright side (despite almost appearing flat 20-12000 Hz).

I tried listening to Supertramp with only the lower frequency cut out, then 2 cuts outs, then all 3 more or less as before.  With each additional EQ cut it sounded better, high frequency fuzz was being removed, revealing voices and percussion better.

Sweeping revealed that the bottom cut at 2729 was already as close as available to a peak at 2736 Hz.  The other two cuts were a bit off from measured peaks, so I moved them closest to the exact peaks similarly.  The 1/2 octave wide and full octave width for the top cut seem psychoacoustically justified.  Frequencies from 6kHz on up sound especially irritating, up to about 12kHz where my sensitivity is beginning to fall.

I'm thinking the Linkwitz-Gundry principle applies all the way up to 12kHz or so.

At first brush, all the upper EQ's seem justified.

For years I dismissed HP's snide comment about the Acoustat 2+2's.  He coined the term "credit card coloration" in reviewing them.  Well, indeed, there seems to be modest peaks at 2736 Hz and a few other high frequencies.  It appears to really enjoy them you need parametric EQ.

(Or at least for me in my listening room.)

Uncorrelated, Full EQ

Correlated, Full EQ

Update May 17

Sometime after making the last measurement, possibly while listening fairly seriously to the Beatles White Album, I realized there was no deep bass on the right side.  In general, deep bass seemed pretty anemic (though it's not terribly in abundance either, in The White Album).  In due course I figured out that the right subwoofer had gotten unplugged.  The IEC pulled out a bit, and I hadn't noticed.  I decided to order a new audiophile cable featuring connectors in solid plastic (though it doesn't appear to be name brand solid plastic, like Marinco for example).  We'll see about those later.

Meanwhile on Tuesday afternoon I knew I first needed to repleat the full spectrum graph that ended yesterday's post, but with both subwoofers operating as they should.  That yielded this:

Both Subs Working

So there's a pretty healthy deep bass, at least between 80 and 31.5 Hz, still nevertheless fitting into my concept of "electrostatic" bass (it's implemented with big subwoofers but thanks to careful nulling of room nodes, it is supposed to sound as if it were very good electrostatic bass).  I ought to be able to fix the weakness about 70 or so since there's lots of EQ there, but not up higher than 125 since that's where the panels take over and I'd have to use boost (again).  Perhaps I shouldn't be so scared of boost now that I'm using right in the center of the midrange.  Below 31 Hz it's damned hard to fix, despite the woofers working very hard, there's massive cancellation in the front of my room where (putting image width and therefore instrumental separation at the top of the reproduction goals) I have my listening seat.  Well it mostly works for me.  Headbangers can sit in the back of the room where they get super exaggerated bass, deep bass included:

To make the base at the serious listening position higher, I'd most likely have to make the bass in the back of the room higher still, and it's already at the wall rattling point.  Except with bass absorption boxes, that would likely have to be about 1/3 the size of the room to have significant effect.  Or a line of active absorbers which cost $1600 apiece.  Or multiple new bass woofers hooked up in swarm.  Or a 2 dimensional FIR approximation of bass moved to the listening position, which might be super-sensitive to where you are listening.

So anyway, that's for the future, I'm still investigating the midrange and upper midrange today.

So let's look at the response with and without the "Gundry-Linkwitz" inspired upper midrange EQ, on and off (I'm going to repeat the graph at the top of this update simply to have both graphs together).

With Upper Midrange EQ

Without Upper Midrange EQ

There are lots of curious things about this.  Starting from the fact that the unequalized version has an apparent peak around 4500 Hz, and that peak is vaporized by the EQ, but the EQ doesn't have a cut at 4500, it has cuts at 2700 and 5300.

The unequalized version looks like it ought to be OK, the rounded peak around 4500 being mostly dwarfed by a gradual downturn that looks like it ought to sound fairly soft.  But in fact it sounds bright and harsh as hell, and not even primarily because of the apparent peak at 4500.

Speaking of which, comparing the EQ'd version with the starting point of this now very extended post, it's not clear my slight readjustments of the EQ nulls to local peaks measured with sweeping worked out for the best.  It appears my earlier adjustments (which were made and remade over many years, primarily to make the RTA look good) were more effective at creating a smoother HF rolloff.  I will have to investigate that later.  Meanwhile, I tried nulling out the 4500 bulge with a corresponding octave wide cut at 4500, yielding this:

This looks like it might sound the best of all, it's a mostly equal and gradual slope from above 2k to 20k, but it's barely changed from the unequalized version in sounding bright and harsh.  At least I guessed this because the sound of the pink noise had barely changed from the No Upper Midrange EQ (NUMEQ) condition.  A little bit better than NUMEQ, but it's clear it still needs more EQ.  I've labeled NUMEQ as "Boost" in my DEQ memories, because it still has the +3dB boosts at 858 and 1013 Hz, both 1/3 octave wide, whereas I've labeled the currently fully EQ'd version "BG" for "Boost plus Gundry Dip EQ."

Perhaps it needs at least one more EQ point to deal with the little bulge roughly 6-7kHz, which is right at the center of the "metallic" sound.

So I added an additional notch just below 7kHz, yielding this (I'm not 100% certain this is right photo but it seems like it has to be)

Well this didn't work out as expected.  Instead of cleanly notching out 7kHz, it's created a new peak above 8kHz and a new dip at 5kHz.  It has a similar rolly polly look as "Boost."  And, curiously, it sounds about equally well balanced as Boost...and maybe better, as a very slight upper nasality has been removed, but it also has too much "brilliance" around 10kHz.  I'm calling this "P2" because it uses two parametric EQ's in the upper midrange.

And I'm calling this next one P3 because it uses 3 parametric EQ's.  This time I went back to correct the resonance right in the middle of the usual Gundry Dip area, thinking that to be more important than the excess brilliance around 10khz.

Now I have a response that looks very similar to BG, but it's constructed entirely differently!  Rather than cuts at roughly 2700, 5300, and 9900 it has cuts at 2700, 4500, and 7000.

Sonically I think P2 sounds like it has more potential than BG, but it's not actually better just different.  And P3 is beginning to sound worse, maybe.

This is all very confusing, but there are several lessons that we know and others we can guess:

1) There is more than one combination of PEQ's that yield roughly the same RTA response.

2) When you whack down one peak, others you may not have been aware of appear.  Also, new dips may appear.

3) We might think that the fewer PEQ's the better.  But this may not be the case.  What we really want are PEQ's that somehow precisely invert "resonances" or combinations of "resonances."  This might mean more PEQ's than at first appear necessary.  But it's hard or impossible to tell where the true resonances are.  With a sine wave sweep, it constantly goes up and down just a tiny bit more here or there.  Without looking at the RTA, I would have never guessed there was a NUMEQ apparent "resonance" at 4500 Hz.  That apparent resonance might actually be the combination of other resonances.

4) Sonically it appears best when the high end above 2kHz is gradually falling.  Rises above this gradually falling floor are undesirable.

Upper midrange PEQ's not right

 I was astonished by the 3 dimensionality of my long time favorite Crime of the Century by Supertramp.  But also thinking the voices slightly disembodied, as if the midrange were sucked out, but you were still hearing the raspiness.

My previously very ad hoc midrange EQ settings, designed to correct some apparent Acoustat peakiness, and yield a "Linkwitz-Gundry" dip, which I thought was pleasing, were:

2729 Hz, 1/3 octave, -2dB

5200 Hz, 1/2 octave, -4.5dB

9038 Hz, 1 octave, -2.5dB

To see how the voices would be affected, I turned all 3 PEQ's off.  The result, somewhat counterintuitively, was less raspiness, and more wholesome sounding voices.

I had actually added the upper PEQ's to combat this kind of problem.  But strangely they weren't helping.

It could be that the factors which lead to some peakiness in the mid highs are not minimum phase, and trying to correct them with EQ messes up, rather than improves, the phase.

I don't actually know what causes the apparent peakiness that I was trying to correct.  I was not following my own guidelines of "correcting by the model not the measurements."  I was just correcting to measurements, and mostly to sometimes highly misleading 1/6 octave RTA.

I'm not going to assume that "no eq" is the way to go.  It simply needs to be done more carefully, including with tests like there where ALL of the PEQ's are turned off, rather than just one or two PEQ's.

For now, I'm back to scratch, with no midrange PEQ's (except the phase corrected crossovers at 125 Hz and 17000 kHz).

Later I was rediscovering Peter Gabriel.  His voice sounded real but just a bit husky.  I bet that's how it was recorded.

I need to set up my ABX to switch PEQ's using midi (as I have done before).  The amplifier test setup is currently dysfunctional anyway (one DAC died...and I also need to get it fixed).  And I have shown over and over that I can't tell the differences between Krell, Hafler, and Aragon amps, though I want to retain the capability of doing that kind of testing.

I also have a weird idea that introducing a small peak around 1kHz might even help.  The original Rogers LS3/5a have a driver with a >10dB peak at 1kHz, which much of the crossover is designed to correct, but perhaps not perfectly.  Perhaps the thing to do is to have a 1kHz peak with an equal and opposite 1kHz null.  What would this do?  Reduce phase information.  That might actually make it sound better.

Vinyl records also have an inherent peak around 1.5kHz I believe, a ubiquitous cutter resonance.

But perhaps also the Acoustats have that very sort of resonance also, that actually compensates for the overdamped nature of the system, and trying to correct it to flat isn't right.

I should also try running Acoustats full range w/o subs and supers.  I have the capability of doing that, I think, with a minute of knob turning.

Another weird thing is that from my adjacent room (kitchen) listening position, where I most commonly listen in background, voices and such inherently do sound more wholesome.  It's mainly as you get right up to the Acoustats that they can sound a bit phase-y.

At the same time, a very close listening position can be astonishingly good, and improves the bass somewhat.

Swapping my high back chair for a low back chair did not affect the voice quality much, and though it makes a big difference in measurements it hardly seems worth changing for most listening.  But still worthy of more testing.

It occurs to me that one way of improving the low frequency room modes is to have a FIR filter designed to move the effective origin of the subwoofer to the listening position.  This can be done by creating a model of what the sound is at the listening position from the subwoofer at it's current position, then inverting it.  This wouldn't yield "perfect" response, but the best available response without making other locations worse.

I think that same sort of "best available performance" is the secret of the Trans Nova amplifier design of the Hafler 9300.  Feedback does not even try to correct the power supply sagging, which is maintained by the inherently high transconductance of the mosfets themselves.  Feedback is taken from the gates not the output.  Despite that, every measurable parameter (except max power), even high frequency high power (20kHz, 100W) damping factor, was equal to if not better than the Krell FPB 300, which uses feedback in the whole output stage by itself (hence very wide bandwidth...equivalent to the Hafler).  When feedback is taken from the output, the feedback tries to squeeze more and more power from the power supply making it sag more and more.

Update May 8, 2022

Looking for an LP to play, I discovered an as yet unplayed Extended Version of Purple Rain by Prince.  It sounded way way way too aggressive in the highs.  Also very thin as if no midrange.

So I tried playing the same thing from Qobuz in high resolution.  I sounded a bit better, but still too aggressive.

So I dialed in the old PEQ's, just as I had them before.

That fixed it.  It's a very hot sounding album, but still listenable with the PEQ's I had dialed in previously. 

It could be the PEQ's are fully perfected yet, but it appears that having them all on is better than all off.

Worthy of Further Investigation

 I've never enjoyed my audio system as much.  It sounds so wonderful, and once clear and also deep, both spacious and holographic.  The latest phase of measurement-based time alignment and phase correction, while still not perfect, is good enough to be a vast improvement over anything I've had before, and almost everything I've heard elsewhere.

And I can and do enjoy it anywhere in my small house, which couples well acoustically to the living room, and an advantage in having a small house.

And I can turn on and off the system from push buttons all over the house.  And now I don't even have to bother picking music, which I could hardly even do once a month, but have a computer program for doing that too.  So I can effortlessly have endless music, wonderful but mostly rarely if ever played before, playing.  The Classical FM station has also been fixed (with my successful prodding) and I often listen to that also, as it's the easiest thing to select via a single push button.

I've been recording albums I haven't heard in decades, and making nice pop-removed (using the Audacity "repair" function, which works perfectly on tiny transients) transcriptions that sound more wonderful than playing the discs live (because I can play back louder than when I made the recording, with no fear, and it's now it's the same exact bits as I would have been listening to then, though lacking the variation on each replay that the phonograph system makes inevitable).

I've tried listening to a playlist of all my "other" music, the music not suitable for background music because it's too loud or has words, and to my pleasant suprise, THAT mostly works as background music too.  I generally don't mind having Beatles or Pink Floyd playing in the background, and it often draws me in for a serious listen.

And I'm accumulating lists of other music to explore, which I can do conveniently through my Tidal and Qobuz subscriptions.

One such list just appeared in the NYTimes, which mentions a lot of interesting artists and musicians.

https://www.nytimes.com/2022/02/10/t-magazine/black-psychedelia.html

Absolute Polarity Revisited

 I am a critic of the idea that absolute polarity is readily audible or particularly important.  A friend of mine has been a leading proponent not only of the idea that absolute polarity is audible, but key to musical enjoyment, and that the recording industry has deliberately reverse-polarized 92% of all CD's to sucker us to buy more and more with less satisfaction.

I've long cited the unsurpassed Lipschutz double blind testing published in JAES in the 1980's.  Lipshutz showed that polarity is not audible in complex music period, but that it is audible on test tones and certain solo instruments like trombone using headphones more easily or speakers in some cases (and Lipshutz used the speakers with unmatched phase coherence, near perfect square wave reproduction, the Quad ESL-63).  However, Lipshutz said it normally costs little to keep polarity correct, so that should be done when possible.  (My friend discounted this research immediately.  "Electrostatics can't do bass, and if you can't do the bass, polarity is moot."  Even though he mostly found the effects of polarity in the midrange and highs.)

The lack of audibility of absolute polarity seems to also be the prevailing view among audiophiles I have met (other than friends of my aforementioned friend) so much that few even bother to get the absolute polarity correct when connecting speakers.  If the speakers are out-of-phase, they simply reverse one to match.  (I always track down where the error is and correct it, to preserve absolute polarity.  I also use an up-to-date acoustic testing app and manufacturer provided audio files to verify the polarity on systems when I've completed setup.  Or sometimes my own constructed audio impulse signals and an oscilloscope.)

I have conducted 3 double blind tests on my polarity-theorist friend, and he has failed to reject null hypothesis p < 0.05 on every one.  All three were of a designs (changed each time, often taking years to construct) he was certain he could pass, as in piece of cake (he was going to market a polarity test kit).  None of this altered his POV of course (it would be plausible to continue to maintain that polarity is somewhat audible, but not so much that it's obvious and therefore a big deal).  Nor does the fact that he has sometimes changed his strong convictions about which recordings are correct (less so with equipment).  Most of his judgements are subjective, though he sometimes also uses (incorrectly, I have many times explained) a schlocky polarity check gizmo, or "simultaneously" starting CD players.

My friend has very often asked me to test polarity subjectively (I have done so only a few times) and criticized me for buying equipment, such as the Oppo BDP-205, and Emotiva XSP-1, which lack a absolute polarity reversal switch.  For a long time he only purchased CD players which had absolute polarity reversal switches for single ended (which are extremely rare, like a few NEC models).  He also criticized many of my purchases as having reverse polarity by default (I tested them and they didn't, then he said they were tricked out to pass polarity correct for test signals only).  To date, the only device out of 100 I have objectively tested having reverse polarity was an iPod at the headphone jacks.

If such switches even exist on CD players they don't control the digital outputs, which is all I ever use.

It has infuriated me that neither the Emotiva XSP-1 nor the exceedingly flexible (but now discontinued) Behringer DEQ 2496 have ways to control absolute polarity either.  I believe there is such control in the miniDSP OpenDRC, but only if you connect a computer device to the miniDSP...there is only one physical rotary selector switch on the OpenDRC, which is used to control "settings" (you could make two otherwise identical "settings" with only polarity being different--what a hassle--but I use the "settings" differently than that anyway).

But I have long, long, long, overlooked the fact that I CAN control absolute polarity using the Tact 2.0 RCS preamps I use.  I just have to press the menu button, scroll to the polarity control, and then the polarity screen comes up.  From there, I can set the polarity of each speaker independently.  This makes instant A/B testing impossible, which is why I have long, long, long overlooked it.  When I first got the Tact I was hoping to do the polarity experiments my friend was asking me to do right away, but finding that I could not do instant A/B tests, I lost interested, and disparaged the Tact as having "no absolute polarity control."

But yet it does have absolute polarity control, it's just not "instant."  It takes about 5 seconds to do the operation.

By that point, 5 seconds on the music has moved on, and is not directly comparable to the preceding music.  Plus your short term direct audio memory is lost, all you have left is whatever memes you encoded from the music previously.  The comparison is far harder...and virtually certain to produce false positives if you believe you must be able to hear differences.  This is even more true if you play something all the way through, or just a few dozen seconds, in each condition.

But this is the type of thing "serious" audiophiles often do anyway.  Including gray hat near-objectivists like me.

So I finally got around to playing with the Tact polarity control on New Year's Eve 2021.  And my first subjective result was that being out-of-absolute-polarity sounded a trifle smoother.  I might add that I am very skeptical I could hear this effect double blind.  It's probably an artifact of the fact that I can't switch instantly.

Other than the likely null effect I am glorifying, I theorize something very different from my friend.  In fact, it was one of many reverse arguments I made to him.  I suggested his all-cone speaker system had magnetic distortion that caused more distortion when transients were played one way vs the other. 

Notably, my friend did all of his actual double blind tests using headphones, which he considered most sensitive to everything...the best window on the recording.  But it may well be that the headphones didn't have the asymmetrical distortion that his speakers may have had which made him believe so firmly in the importance of absolute polarity in the first place.  However, he no longer listens to speakers but only headphones and continues to maintain the uppermost importance of absolute polarity and that most recordings get it wrong.

I don't have an all-cone system...but I do have a cone subwoofer going up to 125 Hz (crossed with linear phase LR8 implemented with FIR DSP).  So, the woofer is important, and it's magnetic, and years of leading transients which tend to do one way could (I imagine) cause a slight magnetization inducing increased distortion.  I imagine the distortion would tend to be reduced, and eventually undone, using reverse polarization.

So, taking my position that reverse polarization will diminish distortion immediately, and in the long run de-magnetize the woofer system, I've decided to try running my system in reverse polarity for awhile.

If I'm wrong, well polarity didn't really matter anyway.  And if that is wrong, and my friend is right, 92% of CD's will sound better anyway.  He taking this very seriously for over a decade now, and he finds reverse polarity nearly always sounds better.

My theory, that dynamic speaker magnetization causing distortion is involved, is objectively testable.  I am doubtful I will find it, though I'll add that to the list of experiments I want to do in 2022.

According to my theory I will continue to get lower distortion in reverse polarity until I have operated the subs for the same number of years in each absolute polarity.  Then I should reverse them again.

Come to think of it, electrostatics could have a long term "memory" distortion effect too,  in both the transformers and the diaphragms too.

 

Audio Achievements for 2021

 1.  I programmed an automatic playlist generator, now called "mplay" (but possibly renamed for distribution so as not to be confused with other things with same name).  I am now planning to "release" it to the public within a few months, I just barely started thinking about a release package in the last week of 2021.  It is likely to be useful only to computer geeks like me who like music.

This has made programming background music from my wonderful album collection a snap.  It also means my vast collection (800 albums and more being added all the time) is actually being used, and not just my top 10 favorite albums.

I was working on this on and off during 2021.  In the last week of 2021 I finally got around to fixing some major bugs.  It actually works as intended now, I think, for the most part.  Previously it had a tendency to keep getting stuck (and never finishing), ignoring all my latest albums (which I hadn't realized), and repeating the same things far more than intended (which I long suspected, but I worried would it be hard to figure out and fix, but in fact only took a few hours of concentrated effort).

I never thought I would do such a thing.  I now wonder how I ever got along without it.  Why did I bother collecting so many albums I hardly ever actually played?

2.  Improving the home automation control of the living room stereo.  I already bragged about this last year, but the job had not been finished.  I added home automation control of the Oppo (just in time for use by mplay generated playlists).  Considerable work was also done making the programming and reprogramming the home control system to make control of the audio system more convenient.

3.  Keeping the local Classical Music FM station sounding good.  Twice this year KPAC developed audio problems.  I reported them and they got fixed.  The first time, the problems didn't get fixed for months.  The second time, in late December, they got the problem fixed in less than a week after I reported it, and I even got a friendly call from the station engineer.  The audio problems were identical on both FM and streaming.  The station engineer said it was a network problem.

4.  Tracking down a peculiar distortion problem.  I figured out that sometimes Sonos attempts to add digital gain when it shouldn't.  A Sonos Update appears to have fixed the problem.  But I mostly use the Oppo nowadays anyway, except for testing, and when I do testing I'll be aware that this issue could pop up again, though I'm taking some measures (like not fiddling with the volume change buttons on the Sonos Node) to prevent it.  This problem might or might not affect some earlier results, such as the miniDSP seeming to need 6dB extra headroom.  I will be aware of that next time I do serious testing.

5.  Fixing a low level buzz.  I discovered I had a buzz about -55dB from normal levels, or -80dB from peak levels.  This was fixed by cutting the shield and ground pin on the AES digital cable feeding the Emotiva DAC.  Funny I had just written a post arguing that audiophiles shouldn't be much concerned about -110dB distortion potentially being added by jitter or other digital anomalies, while often ignoring -60dB noise and distortion.  It turned out that I was yet another person ignoring a -60dB noise.  The fact that this made no difference when I was comparing Hafler and Krell a few years ago suggests that if you can't hear a noise at the listening position, it probably doesn't matter much.

6.  Testing and adding a new digital recorder.  My biggest audio purchase this year was a brand new Tascam DA-3000 digital recorder, for making digital transcriptions of vinyl records and FM radio.  I tested this to be certain it is bit perfect when used with the Lavry AD10 analog to digital converter, and with a clock cable.  Without the clock cable, it is not quite bit perfect.  Unlike my previous recorder, the Marantz PMD-580, the DA-3000 allows the Sample Rate Converter to be turned off, so I am recording the exact bits produced by the Lavry.  I've already recorded a bunch of Sheffield direct-to-disc and Reference Recordings vinyl albums.

7.  Switching to XLD (X Lossless Decoder) for copying CD's to my harddrive.   Up until last year I had always used iTunes for copying.  But iTunes does not always produce perfect copies, and I was becoming disturbed by the errors from some CD's I had acquired from an estate sale.   XLD uses the Accuraterip database to ensure accurate rips.  I also find it easier to use since I can place files into a folder of my own choosing.  Before settling on XLD I also tried Poweramp CD ripper, but I didn't like it as much.

8.  Over 100 albums have been added to my harddrive, including new transcriptions from vinyl.  There are many hundreds more where those came from.

9.  I now use Audacity's "repair" effect to remove pops from vinyl transcriptions.

10.  Minor adjustments to the EQ since the last system crossover and EQ tuning work in 2020.  I itended to do much more work but never got around to it.  Anyway, the system sounds great, far better than anything I've ever had, mostly thanks to earlier work.

11.  Reworked how my Sonos nodes in living room and kitchen are connected.  I'm back to having two nodes in the living room, one dedicated to making HDCD decoding through the Denon DVD-9000.  I have a labeled switch in the Kitchen which determine which kitchen source plays thought the Sonos Line in.  Previously it seemed that always the wrong thing was connected and it was hard to change.  I changed to unshielded CAT6a cables for the Living Room nodes, eliminating another possible source of hum (as compared with using the shielded cables).

12.  Oppo now connected through Ethernet (with unshielded Cat6a cable).  I hadn't even realized I had the Oppo on wifi only.  That possibly explained a few (but very few) previously inexplicable dropouts.

13.  Upgraded my computer CD/DVD drive to a metal cased Other World Computing unit.  My old LG DVD drive started having problems.

14.  Leveled the turntable stand.  This required removing all the equipment from the stand--no easy job--and adjusting the feet on the stand to make it level, then putting everything back.  Previously I hadn't trusted the "wobbly" feet on the stand.  But once the stand is on it's feet, the wobble goes away.

15.  Read the Linn LP12 Setup Manual.  I am hoping to get my LP12, whose suspension started becoming very erratic in late 2021 (probably after I wrongly but briefly flipped the turntable on edge) fixed soon.  But after trying unsuccessfully to fix it myself, I read the LP12 Setup Manual so I have some basic understanding.  I hope to have my LP12 expert, Mark, over here soon to fix and, and now I'll have a better idea of what he is doing.  I tried, unsuccessfully so far, to order a Linn Setup Jig from England.  I obtained a Linn Spindle Oil Cap from a Linn Dealer in USA.  (It's a black OEM one that the dealer claimed to hold tighter than the red original.)

16.  Discovered the source of occasional audio distortion in the bedroom system.  It turns out that I have sometimes set the audio level on the Mac to way over 100% by mistake...this is done simply by moving the scroll wheel while the mouse is over a VLT window.

17.  Got an actual SanDisk interface for reading Compact Flash cards.  This works about 100 times faster than the old Compact Flash reader I had from the 1990's (built into a floppy drive...).  What had been taking 6 hours is now done in 6 minutes.

18.  Copied all my Reference Recordings HRx high resolution discs to my harddrive.  I didn't even realize until recently that I could do this.  These are some of my favorite recordings, and now they are much more accessible as well as being available to my playlist generator.

19.  Labeled all the Surround controls on my Kitchen system.  I use two chifi preamplifiers and two Behringer DEQ's to adjust surround sound on the Kitchen system, and I could never remember which box or control did what.  Now they are all labeled.

20.  Verified acoustic response to 39kHz using M50.  I had a vast array of microphones and systems for measuring up to 40kHz.  Unfortunately, none of them were functioning completely, including my Mighty Mike, which needs a new capsule.  I need to create an usual adapter cable for my Bruel and Kjaer SPL meter which could properly power a few B&K capsules I have, but I've never gotten around to it.

So, in at least a second major purchase in 2021, I purchased my long time dream, an Earthworks M50.  I measured my system using that and my previous standard, a Dayton mike calibrated to 25kHz and compared the measurements.

It turns out my Dayton mike was essentially showing the correct response, with a few minor corrections in the very high end.

The HF curve is:  a dip to 17khz, a peak above 20kHz (the Dayton was showing that peak a bit lower than actual) and ultimately a steep decline in the upper 30's, but not going below the 17kHz level until 39kHz.  (The M50 shows about 4dB more response at 39kHz than the Dayton.)

I think a curve like this is fairly inevitable without a different supertweeter technology, such as my old ELAC 4pi, which could possibly be used now that I can create very steep phase linear crossovers.

Interestingly many of the highly regarded tweeters have peaks like mine in the 20-30kHz region.  Lousy tweeters have peaks in the 10-20kHz region, or steeply rolled off response at 15kHz.  The HF peak helps mitigate the loss of angular dispersion at the high end of a dome tweeter.

Of course, I do use a highly regarded cloth dome tweeter, I think one of (if not "the") best ever made for supertweeter usage, the Dynaudio D21AF, which the manufacturer rated to 40 kHz.  I only wish I'd bought a bunch more of these in the mid 1980's when I had the chance.

But the magic is also in my crossovers, etc.

Dynaudio's response curve doesn't show any peaks, however.  I think that's because it's an anechoic on-axis measurement.

Setbacks

1.  The Linn LP12 developed a terrible suspension wiggle.  I hope to have this fixed soon, but meanwhile I've been using the Mitsubishi LT-30, which isn't too bad I think, and this has also given me the opportunity to make transcriptions of 45 RPM records, which especially includes the ones from Reference Recordings.  I should also be fixing some of my other turntables.

2.  One of the two Emotiva Stealth DC Dacs used for amplifiers powering the Acoustat speakers failed.  This means I can no longer do A/B amplifier comparisons without changing cables.  I also hope to get this fixed soon.

Conclusion

Overall another excellent year for playing music and doing Audio Investigations and Improvements!

Fixing a very tiny buzz

Every now and then I put my ear up to the Acoustats to see if a tiny buzz has been created by recent changes.  Sometimes I think I hear a sound but most often it's the air conditioning compressor (either mine or my neighbor's which is just outside the living room window), the refrigerator, or something else.  I pride myself nowadays on having a very quiet system.  Despite it's complexity.  This is partly because of design, and partly hard work like I just did today and yesterday. 

It wasn't always that way.  For 30 years I just took a small amount of hum and noise for granted.  Most often I could hear a quiet hum distinctly from the listening position with nothing playing.  Then around 2000--my 31st year of audio foolery--I finally started taking hum and noise seriously, and I strangely found that the biggest source of hum in my system then was not my MC225 tube amplifier, which I had long suspected of needing a refurb, but my pristine looking solid state 4 way analog crossover, the highly esteemed Pioneer "Series Twenty" D23.   Apparently because of a failing power supply, it was injecting hum directly into the audio signal.  It was not, that time, a ground loop.   I never fixed the D23, instead I tried many other approaches to doing speaker crossovers, and ended up using DSP around 2006 and ever since.  No matter how much DSP you do, and no matter how many DSP processors, there should be no hum and noise added.  Though you might get ground loops.

A few days ago I was thinking this time I was hearing a very tiny buzz, just barely audible.  It didn't sound like the air conditioning compressor or other noises happening right then, and it was clearly at it's loudest right in front of the Acoustats.  It wasn't easy to be sure, for awhile I thought it was just my hair rubbing against the Acoustat "socks".  I went back and forth between the speakers hearing something similar in both.  I was thinking to myself this could be the Acoustat power supply failing.  So, I shut off the Hafler 9300 amplifier.  It was pretty clear this made the tiny buzz go away.

I then though of doing something I hadn't done in awhile.  I hooked my Fluke 8060 A DVM to the Acoustat terminal with nothing playing.  For a lot of reasons, the Fluke is not necessarily an optimal way to measure audio signals.  But it's very convenient.

I was shocked to see 2.23 mV on the meter, way more than I would have expected.  

So then I put shorting plugs into the Hafler 9300, and measured 0.00 mV.  The Hafler not only was not at fault, it is one quiet amplifier.

So then I tried muting the Emotiva Stealth DC Dac driving the Hafler.  The buzz measured barely different.  While it was still muted, I pulled out the XLR connector for the AES/EBU digital signal.  And then again the noise measured 0.00 mV.

Clearly this noise is a ground loop from the source of the digital signal to the Emotiva.  I was immediately thinking I knew the Emotiva wasn't perfect wrt AES input.  I was thinking AES should not have a hum and noise problem because it's balanced, but the Emotiva implementation isn't very good.  I'm not sure if it uses a transformer.  I remember maybe reading a review which complained about something like that.  (But the Emotiva's AES transformer or lack of transformer is not actually at fault I concluded later, as I will explain.)

I started thinking I should buy a better DAC.  (And perhaps I should.)   But meanwhile I could switch to one of the DACs I have in storage, including an Audio GD Dac 17, and an available Denon DVD 9000.  As far as the Audio GD, I need to do some retesting on it, after a wrongly blamed it for causing the Krell FPB 300 to shut down frequently.  I want to be sure I know how good or bad the Audio GD is before putting back online.  Meanwhile if I used the Denon or most other DACs I have in storage I'd need to convert the AES signal to Coax.  I have several converters for that, but do I know if THEY are isolated or will just propagate the ground loop via the shield of the Coax?  With any of these converters, or even just the Emotiva itself, I might be tempted to use Toslink, which would certainly eliminate the ground loop problem.  So, if I'm going to use Toslink with some other DAC, why don't I just use Toslink with the Emotiva DAC that's already set up?

I tried Toslink and indeed it reduced the noise down to 0.00 mV with the Emotiva muted.

But I don't like Toslink.  I think it's a weak connection very much more subject to jitter than coax or AES.  So after reading online about ground loop issues with AES connections, I decided to do something radical that several people suggested, except perhaps not the way they would do it.  I took the 3 foot AES cable I had been using, and cut out the shield for about 1/2 at the end using wire strippers and cutters.  Along with the shield, there was a drain wire I cut too.  I patched it up with lots and lots of white electrical tape.

Sure enough, this also reduced the noise down to 0.00 mV with the Emotiva muted.  And it worked just fine apparently playing music.

Ground loop fixed!  It rarely gets this good, noise going down from 2.3mV (horrible) to 0.00mV (perfect, so good it's unreal) with one little change costing nothing.

Then with nothing playing, and in fact the Tact preamp itself was muted, I unmuted the Emotiva.  I was shocked to see the noise rise up to around 1 mV (sometimes as high as 1.5mV).  It was still silent at the speakers, but it almost seemed like 1/2 of the measured noise had come back.

Perhaps this was because I hadn't cut the #1 wire also, I wondered.  (Actually, the shield and drain wire carry the #1 pin wire, so in fact I had cut those as well, though possibly not a connection between the Emotiva chassis and the shell of the XLR connector.)

I tried the Toslink unmuted and sure enough, it had slightly more of this noise voltage, over 1.6mV.  That a Toslink connection also had this noise, with no XLR connected to the Emotiva, proves that the noise is not being "introduced" by my having cut the shield on the cable.  The noise appears to be related to the digital signal itself interacting with the Emotiva.

I now figured this was probably very high frequency noise generated by the DAC, perhaps caused by the ASRC, even for the supposedly muted signal.  Perhaps the Toslink connection had even more because it added jitter on top of jitter.

I was still uncomfortable about this new measured noise the next day and I decided to do some more tests.  I bypassed the ASRC containing miniDSP units, and the Crystal 8620 containing Behringers (I'm not sure if they enable ASRC or not) and plugged the DAC straight into the muted (and with no signal playing on anything either) Tact 2.0 RCS preamp, which uses fully synchronous digital receiving and transmission.  The spurious noise was about the same.  So it was not being "caused" by the ASRC's.

But I still really wanted to be sure this was not hum or buzz or any audible noise like that, even if it might not be audible simply because it was much lower than before (and it was barely audible then).

So I got out my Meguro Noise Meter, which has an A Weighting filter enabled.

I ran these tests with the shield-broken XLR cable.  With the Emotiva muted, I measured 0.076mV at the output of the Hafler.  With the Emotiva unmuted, I measured 0.13mV on the Meguro.

So the A weighted noise does not show the dramatic increase to 1.5mV shown in the Fluke's semi-wideband measurement.  This proves in fact the noise remaining, after fixing the ground loop issue, is essentially inaudible.  It is likely to be supersonic noise.

I can see some strange noise on the spectrum display of the Behringer DEQ that does midrange EQ after a miniDSP does the crossover.  Even when the Tact is muted, there is noise around -140dB or so shown on the Behringer display.  THAT may be the effect of dither and/or the ASRC in the miniDSP's.  But the ultrasonic noise we are measuring is much higher than -140dB, compared to the full output of the amplifier (around 34 volts) it is -106dB down.

So it's probably mostly above 20kHz, the cutoff on the Behringer spectrum display.

Not much to worry about.  The big thing this time was the ground loop which was easily fixed.  I grabbed another XLR cable, without the broken shield, and measured noise at 1.8mV on the Meguro.

I got the same results with Toslink and AES disconnected, so once again it's not my broken shield, though for that I repeated the Toslink only connection with the Fluke, and consistently measured just over 1.6mV, about the same as my Coax connection.

BUT, wait...I tried having no input connected to the Emotiva at all, and still measured the same everything...

Apparently this is just the wideband noise of the Emotiva DAC when it is operating, and it is slightly quieter in audible noise when it is muted, but way quieter in ultrasonic noise.  Here I should add that I have the Emotiva gain control set to +2.5dB.  The Gain of the Hafler 9300 is 29.  So the noise appears to be about 0.0016V / (29 * 1.4), which would only be 94dB below 2V.  This seems rather high, though I've never paid much attention to the "Unweighted" noise specs of DAC's if I've ever seen them at all.  Virtually all of the specifications we read are "A weighted," which makes most everything sound better than it is.  The weighted measurement on the Meguro is just over 20dB better, which would be a bit above 114dB, which is a bit below spec.

It's strange, very strange to me, that the DAC has way more wideband noise than my amplifier.  Way more audible frequency noise too, though so low in level it's hardly audible in practice.  I would have never expected this.  I wonder if this DAC is failing.  I already had the Emotiva DAC for one living room amplifier fail (so I can't run ABX amplifier tests right now, I have to disconnect input cables to switch amplifiers).  I'm going to have to look at other DACs.  But the audible portion of this noise problem appears miniscule, so it's not a high priority as was fixing the ground loop.  Back when I didn't care about hum and noise as much, I had the opinion that a little bit of hum and noise might actually make things sound better.  That's still a possibility I haven't fairly tested.  But now I presume that noise should be reduced as much as easily possible (without doing something that might increase distortion, etc).

It occurs to me that whenever you have a DAC with an AES connection, and outputs to coax audio, it is certain to propagate a ground loop.  The ground of the coax output is going to be referenced to the shield ground of the incoming AES.  It therefore makes sense to me that Pin 1 should be completely disconnected on AES devices having coaxial output.  AND the XLR shell should be isolated as well.

Or, maybe it's just that the Emotiva isn't grounded to it's AC input.  Perhaps gear with AES input MUST BE grounded, so that there is a separate chassis ground to which the AES shield is ultimately referenced, which is barely in contact with the active circuitry except perhaps through a small capacitor and/or resistor.  

The other possible solution is to have a cable (like mine now) with the shield and ground connections completely broken.  Rather than crudely strip off the shielding as I did, it would be better to make the cable this way from the beginning.  But so far as I can see, nobody sells a shield-broken AES cable, only a pin 1 lifter (which might not work if the shells are connected).  The shield should be broken on the male XLR pin side (outgoing signal).

Meanwhile, the importance of the tiny buzz I fixed, which could be faintly heard with ear right up to the Acoustat socks, is seriously questioned by the following line of reasoning.  This buzz problem that I just noticed and fixed must always have existed to a greater or lesser degree when I used an AES connection to the Emotiva to a single ended amplifier connection.  That means when I was comparing the Krell FPB 300 and the Hafler 9300 and in blind level matched ABX tests, and found no audible differences in music, I was comparing an amp which had this problem (the 9300) and an amp which didn't, because of its balanced input connections (the Krell).  If little buzzes like this make a difference at all, at least THAT should have been audible in the blind testing.  But it wasn't.

So, this would seem to show that faintly audible noises that you can just barely hear with an ear up against the speaker, are not important in ordinary listening.  The buzz measured approximately 2mV at the speaker terminals.  The rated output of the amplifier is about 34V.  Thats about 85dB down from peak level, or about 55dB down from soft passages.

What does that say about noises that are -110dB or -140dB ?  Perhaps not even worth thinking about ever.

Bit Perfect And Not

 SPDIF digital signals are almost universally decoded nowadays by a certain class of digital input receiver known as Asynchronous Rate Converter (ASRC).  These are most often integrated circuit chips, such as the Crystal Semiconductor CS 8420 (that particular chip is widely used but has a well known bug which is worked-around in higher quality equipment).

These make sense inside DACs as the best-possible way of handling jitter (source jitter, and carrier jitter) from the digital source clock and SPDIF itself.

Assume you have a DAC with a near perfect clock, but what you are receiving is a SPDIF input signal that contains jitter.  Even if the digital source had an absolutely perfect clock, the SPDIF signal from it will contain what I am calling carrier jitter because the actual clock transitions, embedded in the signal, are affected by the signal itself.  The ever varying digital signal itself will microscopically shift the "zero crossing" point enough to affect the determination of the precise clock--where the transitions occur--by enough to produce just below 200 ps of jitter.  This is the price paid for not having a separate signal for the clock itself--which would not be contaminated by the signal.  But note that all serial interfaces in the computer realm have this problem, and parallel interfaces which did not have virtually disappeared...because it's much cheaper to make serial interfaces that are good enough.

If all you have is 200 ps jitter, that is so benign you might as well leave it be.  But source clocks themselves are never perfect, so there will be additional jitter from that, and any mismatch in the speed of the source and receiver clocks will cause sample time units to be gained or lost--and that is not good.

Much has been made of the jitter issue in the subjectivist audiophile press, despite lack of evidence that jitter performance in decent equipment is audible in double blind testing, and much reason (including a major AES investigative report) to believe it would not be.  Often detailed graphs of the distortion sidebands down to -160dB resulting from jitter are shown, and people obsess over jitter sideband peaks which sometimes reach -110dB.  Meanwhile sometimes the same people may brush aside THD+N as high as -50dB (lovers of SET's for example) or -10dB aliases just above 20kHz (lovers of NOS for example) that can and do intermodulate downwards causing massive modulated noise.  If the noise process you are concerned about only causes noise peaks at -110dB, you should not be much worried about it.  Strangely, one of the best sounding DACs to me, the Denon DVD-9000 (which uses dual differential Burr Brown 1704's with an "AL24" digital filter including HDCD digital operators) has a lot of hashy looking sidebands above 10kHz starting just below -110dB and mostly just below -120dB.  Most likely those neither contribute to nor detract from the good sound.  And even those are likely not caused by the synchronous (I believe) digital interface but by the BB 1704's (the best R2R chip ever made, but not as low noise as the best Sigma Delta chips not too long after) and the fancy Denon-proprietary digital filter.  Does R2R have a fundamental advantage?  Well first it should be said the 1704's were quite good and better than the mainstream sigma delta chips for some time in ordinary SINAD measurements.  But secondly, the exact dynamic performance benefits, if any, of R2R chips may need a different kinds of analysis than frequency spectrum analysis to be apparent.  I've drawn a blank on that myself and all the converters I use on a daily basis are very low noise and distortion Sigma Delta dacs--which do sound good to me.  But I'm keeping the DVD-9000's and maybe more just for future tests....and currently for HDCD decoding as well  There are two reasons I'm not rolling DACs anymore.  I've been very happy with the Emotive Stealth DC-1 sonically--very pure sounding--and convenient size, price, and adjustability.  My system needed 3 identical DACs until last year.  Since adding the miniDSP's which convert everything to 48kHz or 96kHz for the supertweeters, the time delay will now stay fixed at different INPUT sampling rates, so I can more easily roll DACs again.  But I don't think it's as rewarding as speaker testing and adjusting.).

The landmark study published in the AES concluded that jitter would have to exceed 10,000 ps to be audible.

Anyway, jitter paranoia has guided design of digital interface receiver chips since the early 1990's.  And it was concluded around then that the best way to handle incoming jitter was to interpolate between the incoming digital values as they come in.  In this way a precise clock at the receiver gets values from the digital interface, but they are not the original numbers from the source, they are numbers interpolated from the input but at the new clock instants.  It's like you have a little computer examining the input values, and educated-guessing what the values would be at the new clock instants.  A fringe benefit of this approach is that it could intrinsically change from one sampling rate to another.  You could extract either a higher sampling rate or a lower one, just by asking the little computer for the guessed values more or less frequently.  This kind of digital input receiver is therefore known as an ASRC.

This approach provided lower distortion than the approach of using a sloppy Phase Locked Loop (PLL) like those used previously to "lock on" to the clock of the incoming digital signal with a small buffer for the digital values.  In that approach, the clock of the DAC itself would be made to approximate the incoming clock, but somewhat smoothed (to remove as much jitter as reasonably possible, including carrier jitter) but follows the clock embedded in the signal close enough so that buffer overrun or underrun never occurs.  So you have to speed up and slow down the clock of the DAC itself, which is hard to do without adding more distortion.

I was long skeptical of the benefit of ASRC until I measured it myself with my Emotive Stealth DC-1 DACS, which allows you to select either the PLL mode (called "Synchronous") and the ASRC mode (called SRC).  The default is SRC and when you select Synchronous, distortion rises from 0.0003% to 0.0004%.  I wouldn't lose any sleep over this, but it proved to me the effect is real, and that ASRC's are pretty damned good.

But this simplistic view does not account for the possibility that not all digital audio devices are either sources or DACs.  What about digital volume controls, EQ processors, Crossovers, Dynamic Filters, Displays, Limiters, and Storage Devices?

When any ASRC is used for the digital inputs of these devices, the values they are starting from on are not the original values.  The are Bit Perfect Not.

If you had a long chain of such devices, all set to "flat" or no change, the digital values passed through the system would be changed by each one cumulatively.  So each one may add only 0.0003% distortion+noise to the signal, but it keeps on adding up.

And if you are storing a digital signal, there is simply nothing better to do than store the original values in it.  Anything else is second (or third, etc) best.  The original values are the best values, having zero noise+distortion added to them, and no process which transforms those values can achieve or beat that.  Meanwhile we don't care if the digital storage process takes slightly more or less time on the order of nanoseconds or more.  It can wait, at zero cost in performance, for each value to come in, whenever it comes in.

My experiments indicate that ASRC's may have other even more insidious problems.  There is a potential for overload from inter-sample-overs.  The best-guessed new values between digital sample values can actually be above 0dB in some instances, especially with extremely highly compressed recordings.  Inter-sample-overs can be as much as +6dB which is severe digital clipping.  To be sure you will always avoid this problem, each digital processor must lower the digital signal going through it by 6dB.  This means the dynamic range keeps dropping by 6dB for each digital stage having an ASRC input.  (The presence of inter-sample-overs shows that the interpolation method used by ASRC's is not the linear interpolation we learned in High School, but a higher order interpolation method.)

This is little problem in a DAC where you can have more than the carrier number of bits operating inside the DAC itself.  You can have 32 bits for an incoming 24 bit signal, giving you way more dynamic range than needed.  In effect you have headroom above the headroom of the carrier signal.

But it is a huge issue for a chain of SPDIF connected DSP and storage devices (for which, BTW, the carrier jitter stays roughly the same no matter how may SPDIF interfaces are in the chain...at the end of 10 devices I still measure about 200ps jitter--same as from the input--because that's just inherent to the SPDIF carrier itself (as described above) and each synchronous interface reduces it internally to near zero with a PLL, then it goes back up to 200ps at each SPDIF output because of the SPDIF carrier itself...which all goes to show how much of an over hyped concern jitter is).  Each one is going to reduce the potential dynamic range by 6dB because it has to output back into the 24 bit domain.

Fortunately, many of my devices are old school synchronous SPDIF.  That especially includes my 2000 vintage Tact digital receiver, which even boasts about locking on to the clock of any of its digital inputs.  And it's true of my vast army of Behringer DEQ 2496's, each an extremely flexible DSP and Display device which is sadly now discontinued.  And, especially, my Alesis Masterlink, which records the exact digital values send to it, thanks to using a synchronous digital interface (CS 8416 I think). 

SADLY, most new digital processing and storage devices do NOT have synchronous interfaces for SPDIF.  This includes the fairly ubiquitous (as a replacement for Alesis Masterlink) Marantz PMD-580.  The Marantz uses an ASRC to accept whatever digital signal is provided, from 32kHz to over 96kHz, and convert it to whatever rate you choose to record at, maximum 48kHz.  There is no way to turn this "feature" off, and that is typical of digital recorders you see nowadays.

The TASCAM DA-3000, their current 2 channel flagship, does allow you to turn the SRC on or off.   When I saw that, I knew I had to have this unit to replace my extremely cumbersome Alesis Masterlink, and my ASRC-centric Marantz PMD-580,  to make digital recordings.  (I use my Lavry AD10 as analog-to-digital converter, and I suspect it may still be better than the DA-3000--though limited to 96kHz which is fine by me--and then pipe the AES/SPDIF signal to the digital recorder.)

It took considerable time for me to figure out a way of confirming that the DA-3000 makes bit perfect recordings.  And then my first measurements...which suggested it did not...were incorrect.  Finally I have concluded it does make bit perfect recordings at 48kHz when fed that signal from the Marantz PMD-580.

But with one strange caveat.  Once every minute or so there is an extra sample added or subtracted. 

I figure now that is because the "Synchronous" option works best when you also use a separate word clock signal.  I have ordered a Word Clock cable to check out this theory.  If that is the explanation, I'm good.

Another possibility is that this extra sample being added or subtracted is some kind of watermark.  I suspect it's not audible, but I would not be happy about it.

Using the same test procedure, I found the Alesis Masterlink is indeed Bit Perfect, though it might take one sample or so for it to lock on to a new signal.  (The DA-3000 takes 7-14 samples to lock on.)  The Alesis neither has nor requires a clock input or output.  At the 48kHz sampling rate, it just works perfectly without one.

Method

I recorded 30 seconds of music from the FM radio recorded on the Marantz PMD-580 at 48kHz.  I then played this over and over through 18 feet of Belden coax (2 pieces joined with an gold RCA double barrel...this is the "line" I have always used to play the PMD-580 on my system...but obviously not "ideal") into different digital recorders also at 48kHz, 24 bits.  I transferred the files (either on CD24 or CF) to my Mac.  There I used Sox to trim the leading zeros, and trim the end to the maximum shared length, and then to convert to a .DAT file having all the samples as numbers in a DOS text file.  Using the text editor Emacs I edit this file in several ways.  Primarily I edit to remove the column of time numbers which differ slightly between runs.  Just after the "signal" is starting in the digital recording, I look for the first matching line (pairs of values representing the sample value in each channel) in the two files being compared.  I edit out the earlier lines which result from one recording not starting as early or as fast as the other (depending on how fast the recorder locks on to the digital signal).  Then I also edit out mismatching lines after they both go back to all zeroes as that ending segment will vary depending on how fast the recording was stopped.  I then compare two files (typically both from the same recorder) using "diff" (terminal command in bash) to see any mismatches.

I have not yet tried using a word clock cable, but the current results suggest it is necessary for the DA-3000.

Results

For the Masterlink, once an initial sample or so is removed, two separate recordings match perfectly.

For the DA-3000--and with SRC turned off, once 7-14 samples are removed from the beginning of one recording or another (because of slow locking) they match perfectly, Except for about 1 sample added or subtracted every minute.  

I could not and did not need to the Marantz PMD-580 digital recorder because the SRC cannot be turned off and so it never records the original values but always interpolated values.  The specs for the AD converters aren't very good (and don't sound good to me) but no specs are given for the ASRC and I think it's fine as far as they go.  I haven't yet figured out which chip it uses, but obviously an ASRC chip.  I have never noticed bad sound when recording the output of the Lavry AD10 running at 96kHz converted to 48kHz by the Marantz, and actually the 48kHz recordings are smaller and more convenient, so it's wonderful for recording FM radio while using a better ADC (like the Black Lion) in front of it.  I'd hope for something better for recording vinyl, and it just bugs me that I cannot in principle make bit perfect recordings with this recorder.  It will always be subtracting something, measurably but likely imperceptibly, with the ASRC, when it's not really needed for recording to a flash card.  Therefore it is never giving a truly honest account of the digital signal, it is always part of the mix.  For the purposes of relaxed listening, that may be ok, but for the purpose of Audio Investigations, it is not acceptable. 

Conclusion

The digital interface receiver of the Alesis Masterlink appears to be the best I have tested, providing perfect results even in the less-than-perfect test setup.  I sure hope the errors with the DA-3000 with the SRC off go away when I add a word clock cable, otherwise I'm returning it.

Chips

Digital interface chips using ASRC

CS 8420 widely used but has well known bugs, must sometimes be restarted, so requires a design with microprocessor that can manage that, despite spec sheet alleging otherwise.  It can occasionally go into "garbage mode" or "muffled mode" and when that happens it needs to be restarted.  Some have characterized it as "evil."  It can be used in one of 9 different modes, including several which are PLL only (no SRC).  So you can give the user a choice.  But the PLL only mode may be inferior to that on actual PLL interface chips--it's just a pre-smoother for the ASRC.  The chip is supposedly discontinued (still widely available it seems) but current designs inspired by it are legion.

I see now that my beloved Behringer DEQ 2496 Ultracurve Pro units use this very chip.  I will have to investigate whether they use a PLL or an ASRC mode.  Somehow it seems they do preserve input sampling rate at the output.  The Ultracurve itself was discontinued in 2021.  I know because I've been ordering 1 or 2 a year.

AD 1896 an early reference standard following from the pioneering  AD 1890

WM 8805

Nowadays some DAC chips have the digital interface built in, and if so, you can bet it's ASRC.  PLL's require more ancilliaries which was why, historically, there was a separate chip receiver.  ASRC makes it possible to make everything cheaper. 

Digital interface chips using only PLL (Synchronous) only

CS 8416, widely used a decade and a half ago and still, has 8 digital inputs, often only 1 used.

CS8414, a slightly inferior predecessor to 8416 (though some say the reverse).

CS8412,  earlier generation, regarded by some as the best of the 84xx series.

AK 4117, possibly very slightly better than 8416 if you can find it.  I'm not sure of the design of later AK interface chips.

https://www.diyaudio.com/forums/digital-source/73446-dac-design-first-step-spdif-receiver-print.html

TI DIR9001  Possibly the best PLL chip of all at least when it was introduced.  People were saying this wouldn't be available after 2005, but it appears that you can still buy it new online.  So, it seems now, the designs and redesigns that were inspired by the widely feared impending demise of this chip were wrongly inspired.  HOWEVER, it is limited to 96kHz digital inputs.   So I say, what's wrong with that?  What high end gear that is not terminal (like a DAC) should do is use a real PLL chip for the PLL modes, not the crappy PLL of a chip mainly designed as ASRC.  And then if PLL doesn't bother with anything above 96kHz, like the great PLL chips of old (and still now apparently) that's fine by me.

The Alesis Masterlink uses AKM chips so it seems likely it uses the AK 4117 receiver (or earlier generation of same) which is said to be better than any in the Crystal Semiconductor CS841x series.  I wonder if the Tact 2.0 RCS uses the TI DIR9001 which is a high performance chip made by an American company like the Tact itself.  I have never directly compared the two but generally they both work well.  I've only rarely used the digital inputs on the Masterlink (unlike the PMD 580 it has decent sounding built in analog to digital converters that I've used instead), or even used it at all (I don't make recordings very often), whereas for 20 years I've endlessly used and tried to do impossible things (such as 100 foot cables) with the Tact, so I know the Tact is slightly less robust in locking to 88.2kHz than at 96kHz with long cables, possibly having nothing to do with the chips used.

TI DIR1703 was an early buggy version of DIR9001 from the mid-to-late 1990's.

[Update December 7, 2021]

Tascam DA-3000 Passes Bit Perfect Test with Clock Cable

I finally had time on December 5th and 6th to retest the Tascam DA-3000 using a clock cable, as I correctly surmised that it needed.

I could not do a repeat of the previous tests, except with the clock cable, as I had originally planned (and why I bought a 15 foot clock cable).  

What made the test impossible is that the Marantz PMD-580 does not have any clock inputs and outputs.  The Marantz is built with the philosophy that an ASRC operating all the time is just dandy.  It is not possible to turn it off.  You never get the exact original bits, but the Marantz meets its noise and distortion specs using ASRC.  For many purposes (except high end audio and audio investigations) that is Good Enough.

So I had to set up an even more elaborate test using the Lavry AD10, which is the important thing for me in actual use.  What I wanted to do all along is want to record the bits from the Lavry AD10, a very respected and nice sounding analog to digital converter which is still being sold as a new product for about 50% more than the DA-3000 itself.  I have always used the Lavry in the Living Room system to digitize vinyl records into my digital front end just for playback.  It was the best analog to digital converter I could afford (and still).

Since I already digitize vinyl simply for playback on my system, I would have thought an inexpensive device could simply capture those bits perfectly from SPDIF and record them to a Compact Flash card (or better yet, a USB memory stick).  Unfortunately such an inexpensive device does not seem to exist.  Inexpensive recorders actually tend to have no digital inputs or outputs at all, and definitely not AES balanced digital inputs and outputs.  Earlier generations of the Tascam had no way to turn off the SRC, just like the Marantz PMD-580.  The only commercial bit-perfect recorder that I know of other than the DA-3000 is the Masterlink ML-9600, which hasn't been made in almost 2 decades and is cumbersome to use, requiring the burning of a CDROM for every data transfer.  It's also too noisy to use in the Living Room, I decided years ago (though I now understand a tricky DIY SSD upgrade is possible, and that is said to eliminate the noise which I previously believed was caused by a non-removable fan).

I did use the Masterlink for all my earlier vinyl transcriptions, before I had vinyl playback in the Living Room.  In the bedroom, I arranged the masterlink so it's vent holes (I figured there was a fan there, but it might be just the harddrive) pointed away from the turntable, or at a different elevation, so it was never much a problem.  In the living room, the noisy vent holes of the Masterlink would be inches away from the turntable, not a good situation.  That was what led me to acquire the Marantz PMD-580.  It was only after I acquired it on eBay that I discovered you could not turn the ASRC off, which is contrary to my goals.

I had figured I'd repeat the PMD-580 to DA-3000 test, but just with a clock cable.  That would be holding everything else constant, except for the clock cable.  And then I'd test copying bits from the Lavry, the real goal, after that because it's a more complicated test.

But I had to skip straight to the more complicated, but also more important (like actual usage) test.

To do that, I connected the DA-3000 clock output to the Lavry clock input, with "Word Clock" selected on the front panel of the Lavry.  When it is receiving an external clock signal it can handle, the corresponding sample rate light lights up.  It is pointless to preselect the sample rate when you are using an external clock.

The AD10 has a Clock Input which accepts either Word or AES clock.  The DA-3000 has word clock inputs and outputs.  I do not need to change the clock setting of the DA-3000 since the work clock output is always active at the sample rate currently selected.

The AES digital output of the Lavry already goes through a Henry Engineering 4 way AES splitter (which operates like a little line amplifier buffer with zero delay) so that I can play on my system while recording, previously on the Marantz.  But now, to record on the Tascam, I ran AES cables from the splitter to both the DA-3000 and the Masterlink ML-9600.

So, whatever I record, I record it identically to both recording devices.  If the recordings turn out to be identical, then they must both be bit perfect.  I already proved the Masterlink is bit perfect, but even if I didn't know that, identical recordings on Masterlink and DA-3000 would prove they were both bit perfect, since they could not match if they weren't.

The pink noise track from the Stereophile Test CD 2 was playing over and over on the Oppo BDP-205, and the balanced analog outputs of the Oppo were feeding the Lavry analog inputs, with the Emotiva XSP-1 in between doing level setting and buffering.  So I'm not recording the actual bits on the CD, I'm recording the bits produced by the Lavry converting the analog signal which originated at the disc player.   Every recording will be different from the previous one, because the recording never starts at the exact same instant of the signal, but if both recorders are bit perfect, the corresponding recordings made from the two machines machine in the same session should match.

I recorded this track 7 times on both the DA-3000 and the Masterlink.  I primed both machines by pressing their record buttons.  (On the masterlink, I initially had to select playlist 1, then playlist edit, then every time I also had to press new track, before pressing record.  As I said, the Masterlink is cumbersome to use.)  When there was a gap in the playback of noise, I pressed Play on both machines (which is what you do to start recording, because the record button only engages "Record Pause").  Then when the track ended, I pressed Stop.  And so on, seven times.  I did it more than once in case I messed something up, and it's easy to keep doing once you get rolling.

As it happened, when I made the transfer CD24 disc on the Masterlink (containing all the 24/96 digital recordings) there had been one song from the previous tests still in the playlist.  I thought to myself "no problem, I'll just remember that when comparing."

As it turned out, by the time I was actually messing with all the files on my Mac, I forgot about that difference in track numbering.  And as a result I spent 30 minutes trying to make two files that would never match (because they were different recordings from a virtual analog source) line up because I could not find matching pairs of numbers.  I got very frustrated and angry.  But then I remembered I had to test a higher number Song from the Masterlink to the one from the Compact Flash which came from the DA-3000.

Once I got the two digital recordings to line up in time, and then removing the time counters from the files, using the method I've described before, they matched perfectly.  40 seconds of digital data at 24/96, almost 4,000,000 24 bit samples for two channels without a single difference.

The terminal on my computer screen looked like this:

(The two pound signs shown after the diff command were my initial faulty attempts to do a screenshot.)

Having passed the test, on the evening of December 6 I removed the Masterlink for storage and "permanently" set up the Tascam DA-3000 as my living room recorder.