The Lavry AD10 Microscope

My 12 year old Lavry AD10, which had been left running 24/7 since I bought it (I won't be doing that any more) died when during some power surges when my home foundation was being repaired.

I sent it to Lavry, who repaired it for $600 by replacing the main board.  So basically I have a brand new Lavry AD10 (which appears to still be in production).  It might even be better than the original unit (purchased in 2010) ever was, but I never did complete measurements of the Lavry before.  I figured I didn't have anything good enough to measure it with.

The Lavry AD10 has a signal to noise specification that appears to be about 4dB better than the AD converters of the Tascam DA-3000 recorder (117dB vs 113dB).  Since the DA-3000 is a newer device, and noise specifications depend heavily on test protocols, I really didn't know which was better.  But I figured the Lavry was probably better, and I was right.  My measurements yesterday suggest the Lavry is indeed just about 4dB quieter than the DA-3000.

I've long said that digital converters are about the best thing made in audio because a lot of attention has been focused on them.   This may be less true of analog to digital converters nowadays (up until about 10 years ago, typical analog to digital converters were better than the digital to analog converters, somewhat counterintuitively, but now consumer DACs can have better than 130dB S/N which is as good as megabuck ADC's).

I've often said they were so good, digital converters are often and generally better than preamps.  Perhaps even my Emotiva XSP-1, which I envision as about the quietest digitally controlled preamp you can get under $10k (above which perhaps Mark Levinson makes even quieter digitally controlled preamps, though FWIW the Emotiva specs are better than all but the most expensive Levinson "Reference" models and about the same as those Reference models, so we're already getting about as good as it gets with the Emotiva XSP-1, in performance anyway).

Digital converters are probably better than nearly all non-digitally controlled preamps too, which in some cases (probably not many) might be still quieter.  But I like the digital controls for setting levels precisely and also ensuring perfect stereo tracking.  I got so frustrated by channel tracking I switched to digital preamps in 2000, my first being a Classe CP-35, not only because of the perfect level tracking but to my ear* it even sounded better than passive and non-digital preamps I had been using before.  I've found no need to go back to non-digital preamps since then.  (I found the XSP-1 to be far better sounding* than even the Classe).

Now I have studied this issue in some detail.  It appears that there is an unimportant flaw in the noise spectrum of the Emotiva which is made invisible by the noise of the DA-3000.  But like a microscope, the Lavry AD10 clearly shows this flaw, an ultra low frequency noise in the left channel only (both of my two Emotiva XSP-1's show this same identical flaw).

It's worth recounting how I came to use the Emotiva XSP-1 as my living room preamp for all "analog" sources (including the analog outputs of DVD-Audio players which preserve the full 24bit resolution while the digital outputs truncate it to 16 bits).  For many years I was using a passive switchbox to switch among such sources, and the level adjustment on the Lavry to set levels for digital encoding for my downstream digital processors.  But the Lavry level adjustment lever, which is great for long term settings, it a pain to reset on every disc.  A big pain.

Then I also discovered that the noise level of the big XSP-1, which I had originally purchased for my less high end bedroom system, was even lower than the tiny XPS-1 phono stage I was using in the Living Room, which itself was lower than my dB Systems high gain preamp (which stunned me, because the dB systems preamp and the XPS-1 and the XSP-1 all use the same low noise preamp chip for phono pre-amplification, but somehow it was about 10dB quieter in the XSP-1 as compared with everything else).

So I needed an XSP-1 in the living room just for the phono preamp alone.  And all my bench measurements of the XSP-1 suggested it was about as perfect as I was able to measure, with S/N better than 110dB and distortion below 0.005%.

So the combination of convenience and performance led me to use the XSP-1 not just for phono preamplification, but also "preamplification" (mostly downward level adjustment) for the special digital sources (DVD-Audio**, SACD, and HDCD) that cannot be output in their full resolution through SPDIF which I need for my digital crossovers and equalizers.

Now using the "Lavry Microscope" I can see a flaw in the XSP-1 more clearly.  This same flaw was invisible amidst the mere 4dB higher noise level of the DA-3000.

I obsessed over this Emotiva flaw for at least one day.  But although I toyed with the idea of using a passive switch rather than a preamp for the digital sources again, I have once again concluded the XSP-1 is "good enough" not to bother with that.  Slightly higher noise levels below 5 Hz (and still below -120dB) are just not that important.  That's probably what the designers of the XSP-1 thought too.

[Pictorial section being expanded.]

First I wanted to re-measure the setup I've been using, Oppo BDP-95 into Emotiva XSP-1 into Tascam DA-3000.  I also measured the Lavry with open inputs (not shorted, an oversight).  By itself, with open inputs, the Tascam had about 1 dB less noise than with the full chain.  That suggests that the Tascam was generating more than half of the noise, which would be about what I'd expect.  

The picture below shows about 5 minutes of recorded noise amplified digitally by Audacity by 94dB.  So we're taking a close up look at the noise, about as close as we can get as there is only about 0.3dB of headroom on the right side, which is the full chain of equipment, whereas just the Tascam by itself is on the right.

Things didn't look so good with the Lavry inserted in front of the Tascam to do the A-D conversion (using the Tascam wordclock signal for synchronizing):

The (top) left channel noise looks "noisier" somehow.  Notice that the right channel noise is significantly less than when recorded directly by the Tascam in the previous picture.  Almost 4dB lower in fact, just as the specs of the two AD converters suggest.  And if you carefully compare the right channel to either of the Tascam direct recordings, it's actually lower, but not as much as the right channel, because of some extra noise that's being added somehow to it.

My first concern, in fact the main reason I was doing these tests was to be sure that the Lavry, just back from an expensive full board replacement, was now working properly.  And this first measurement didn't look good.  And possibly because I go about things in a more round about way than necessary, I didn't fully prove that the Lavry was fine and good until about ten measurements later.  Well I was also concerned about my Emotiva and Oppo which might be just as expensive to repair.

And it turns out that the problem itself is no big deal.  But I've decided to tell this story mostly as I experienced it, so it becomes clear that way.

It seemed to me that the first and easiest thing to do would be to reverse the channels.  The right side of the picture shows the normal connections and the left side shows the reversed connections.

The extra noise in the left channel moves to the right channel after I reverse the cables.  So therefore, whatever it is that is causing the extra noise in the left channel with normal connections must precede those connections.  It possibly comes from the cables themselves, so I then tried putting the Lavry connections back to normal and reversing the output connections at the Emotiva:

After all the messing around with cables I had done just to get the audio XLR cables moved from the Tascam to the Lavry I worried that I might have damaged them.  But it was clear that it made no difference whether the cables were left-to-right reversed at the output or the input, the result was exactly the same.  So the XLR cables between Emotiva and Lavry were exonerated from causing the added noise.  

At this point I decided to "mute" the Emotiva preamp.  I'm not sure whether I dialed the volume all the way down, or turned it off.  (My recollection is that I was going to turn the volume down, but I made the recording without even turning the Emotiva on.  So I called that "muting the preamp" in my too-brief notes.)  The strange extra noise in the left channel went away completely.  From this point on, I was convinced that the problem was not in the Lavry, however I didn't fully verify that until a later test in which I put shorting plugs in the Lavry inputs.

It was at this time that it occurred to me that prior to having the Lavry repaired, I had taken great pains to connect the AC power to the same power strip as the Emotiva and all the front end components.  It looked impossible to make this change without disconnecting and moving heavy equipment like the almost 50 pound Denon DVD-9000 from the rack in order to get at the power strip behind it.  But somehow I managed to get it done anyway.  I even used the same 3 foot SJT power cord I had been using before.

The results seemed like a significant improvement when I was first examining them, but on checking them now, I'd suggest they were not any change at all.  However in either case the extra noise in the Left channel did not go away.  That part had not been fixed.

Then I removed extra cable after extra cable from the preamp in a whole series of tests I won't bother you with here because they were all the same.

Finally I went all the way with this sort of test by removing every single cable from the Emotiva except AC power and the output cables connecting to the Lavry.  I selected a balanced input (2) which was shorted with XLR shorting plugs.  The extra noise did not go away (note that the channels are still reversed because the output cables were reversed at the Emotiva).  The Emotiva left channel (bottom) noise looks different from previous images because now we are looking at just one minute instead of 5 minutes.  After spending hours doing these tests, I decided I could just as well get by with 1 minute recordings.  Now it's becoming clear that the extra noise is a very low frequency baseline shifting around -98dB in level (these pictures are amplified 94dB by Audacity, so a full scale noise would be -94dB down):

Now that I had the XLR shorting plugs out, I plugged them straight into the Lavry.  The result was the lowest noise of all with no extra noise in the left channel whatsoever.  To compute the peak unweighted noise level, I bring up the Amplify dialog one more time, and it shows me how much more amplification is possible.  In this case it is showing 5.78dB more amplification is possible  Since the starting amplification here is 94dB, the signal to peak noise level is 94+5.78  = 99.78dB.  Applying RMS adjustment and A weighting would probably improve this to almost 120dB or maybe better.

I concluded there was something "wrong" with the Emotiva XSP-1 preamp that was causing a small but measurable extra noise in the left channel.

So the next day, I moved the pile of mostly spare equipment away from the side of the rack so I could swap the living room and bedroom XSP-1's.  The living room XSP-1 measured above was purchased in 2018.  The bedroom XSP-1 was purchased in 2014 (it's also the second generation btw) but was repaired (basically refurbed) in 2019 by the factory.  As part of the repair they do a full AudioPrecision test which verifies it meets all specifications.  Since the repair, it has not been used very much (and I was careful to keep it turned off when not in use, something I should have been doing from the beginning because even if nothing else the display gets rather dim in about 5 years of on time).

As it turned out, my other XSP-1 was almost identical to the one measured above.  The added noise in the left channel looked almost (but not entirely) identical.  The noise in both both channels was just over a half dB lower, so I decided to keep this other XSP-1 in the living room from now on (and it won't get unwanted wear now that my home control system turns it the living room preamp on and off using the trigger signal).

It was only now I started to probe the noise itself.  You can see recurrent peaks in the range of 2-3 seconds (see the selected range in the above picture).  That means the noise is in the range of 0.5 Hz to 0.33 Hz.  We're talking about very low frequencies here.  And we're also talking about very low levels too, certainly 97 dB or so below peak level.  That's good, but not as good as the midrange noise floor which seems to reach down to -150dB or so (better than I should be able to measure, though it seems I can).  Here are the spectrum plots of the two channels, first the better one.  Remember to subtract 94dB from the levels shown in the spectrum.

Emotiva Right (better) channel

Emotiva Left (worse) channel

The main difference here is that the noise seems to rise below 4 Hz faster in the left channel.  In both cases the midrange and high frequency noise level is extremely low.  The signal to noise in the midrange is around 154dB (94+60).

This low level ultra low noise (ULF) noise in one channel would not even appear in an "A" weighted noise until you were many decimal places out.  Very low frequencies like this simply aren't audible and usually aren't measured either.  Listening to the noise (amplified by 94dB by Audacity) the left channel with the ULF noise if anything sounds softer and more pleasant than the right.*

Many preamplifiers are going to have input capacitors which completely filter such low frequencies out.  And sooner or later it is almost certain some component will filter it out.  And it's very low in level (-97dB or more) to start with.

A nearly identical very low frequency noise in just the left channel is a flaw in both my two units, including the second which was sent back to the factory for repair 2 years ago and hasn't been used much since.  It looks to me like a design flaw that Emotiva isn't concerned about, and truly it isn't of much importance either.

In fact, it's rather surprising that the Lavry AD 10 has frequency response which extends down to 0.33 Hz, but it appears like it must.  Whereas the Tascam DA-3000 does have about 4dB more noise, as the specs suggest, which partly covers up the Emotiva ULF noise, and partly it may have low frequency filtering which also hides it.

So at the end of the day, the difference seems to be that the Lavry captures the ULF noise of the Emotiva both because the Lavry is so quiet, and also because it has low frequency response that seems to extend to something like DC, whereas the Tascam does not.

But just because you can see something with some kind of measurement, doesn't mean it's important.  I can't imagine a single good reason why such a ULF noise would be important not to have.  At this point, I don't even have any data about whether other preamps have similar issues.  With most tube preamps, noise would overwhelm such small effects, and in my experience with tube preamps they flopped around their DC levels not just in microvolts but in millivolts if not volts.  So there you could say the XSP-1 has a bit of that kind of "tube character" in very attenuated form.

Though my theory is that it's the output servo loop of the left channel being closer to the power supply or computer or something like that.  An issue that could be resolved with yet another board layout revision.  But the XSP-1 is beyond revisions now, it's been discontinued (which I'm not happy about either, I think it is a very fine preamp and I don't believe Emotiva or anyone else has a close enough replacement for me now, though fortunately I don't need a replacement now, my two XSP-1's are working fine, good enough in my opinion--did I mention the midrange noise floor looks to be in the vicinity of -150dB, that's the kind of thing that actually counts).

The next day I decided to test an alternative theory, that the problem was being caused by a ground loop in the (coaxial) clock signal from the Tascam to the Lavry, possibly causing jitter.  (They are plugged into different AC power strips and that is almost unavoidable.)  I found that using the internal clock on the Lavry and the Sample Rate Converter on the DA-3000 made no difference.  (My notes are unclear if I also disconnected the clock, which wasn't entirely necessary for these tests but would rule out a ground loop on the input circuit.)  Also I shorted the left input on the Lavry but using the usual clock cable, which eliminated the extra noise in the left channel.  And remember the channel reversing experiments and shorting experiments above, which were also all done with the usual clock cable.  It seems to be disproved that the clock signal is in any way involved in this extra ULF noise.

Here the left channel is on top, but the Lavry left input is shorted.  The bottom right channel shows a little ULF noise, but unchanged from the right channel in previous measurements and far less than the left channel.

Unhappy that I had failed to note if I had removed the cable or not, I decided to do another followup test this time making sure I removed the cable.  By this time (and also in the previous set of tests) I had already figured out a small optimization.  I reduced attenuation on the Lavry by 3dB so that peak level is now -8 (around 5.5v peak) instead of -11 (3.8v peak).  Then I increased the gain on the Emotiva to +4.   The Emotiva can just as easily handle that balanced output voltage.  This optimization could in theory increase the S/N by 3dB.  (I set the Emotiva level to be just before the level it causes clipping on the J-Dunn test.  Since I reduced the Lavry sensitivity by 3dB I might have just raised the Emotiva to +3, but it now seems like +4 is the correct level to reach closest to 0dB on peak signals.)

The following tests showed that removing the clock cable (which requires setting the Lavry to internal oscillator and enabling the SRC on the Tascam) makes no difference at all to the ULF noise in the Left channel. 

Clock Cable being used, +4dB higher level

Clock Cable disconnected, 4dB higher level

It is looking like the level readjustments may also have reduced the ULF noise from the Emotiva.  The new noise levels (with clock connected as usual) are amazingly good, peak noise -99.2dB in right channel and -98dB in left channel.  A weighted RMS S/N would be in the vicinity of 120dB.  This is the full chain including Oppo, Emotiva, and Lavry.

Right Channel Noise

Left Channel Noise

Now I wanted to see how much the gain changes were making this better, so I went back to the old gain setting on the Lavry (-11dB reference level, ie +11dB gain).  In fact, the gain changes (previous set of measurements) were making a pretty big difference.  The new noise level (with Emotiva at +1 to optimize J-Dunn headroom as was done in last two measurements) is -95.5.  So there has in fact been about a 2.5dB improvement in lowering the ULF noise from the Emotiva by lowering the Lavry gain 3dB.  The other channel noise level has not changed as much, presumably because it's mostly being caused by the Lavry at this point.

Emotiva at +1dB, Lavry at -11

Most of the prior measurements were done with Emotiva gain at +0.  This was artificially making the S/N look "better" than it actually was.  Here is a replication of what I was doing before.  It "looks" like the peak noise level is -95.9 but that is misleading because full scale cannot be reached.

Emotiva at +0, Lavry at -11

Because I'm not boosting the level, I wanted to see how much effect this has on distortion caused by the Emotiva which may be putting out more than 2V balanced into the Lavry.  I used a CD on which I have recorded digitally generated (by Audacity) 880 Hz at maximum level (0dB) but no clipping.  I changed the Lavry gain to 0 for these tests.  I tested this "maximum output" CD at Emotiva gain levels of +0, +4, +9. and +10.  At Emotiva gain level +11 this signal was clipping the Lavry (even at 0dB gain on the Lavry so I could not even make the measurement).

At +10 (which is 6dB higher than my new standard level) there were no visible peaks above the -90dB bottom of the Audacity spectrum (but is this misleading?) and the bin value was -101dB but the peak value was -28.8dB.  That would mean about 4% distortion.  My guess is that the "peak" value is more representative of the harmonic distortion peak but it might be exaggerated.  All I can say for sure is that somewhere above +8 we turn a corner and distortion starts rising. 

Emotiva at +10 with max 880

The very best harmonic distortion measurements were with the Emotiva at +8.  At that point, the peak level of the second harmonic is is -110.6dB, corresponding to THD of 0.0003% or better.

Backing the Emotiva down to +4dB gain, the distortion rises slightly to -107.7, or about 0.0006%.

Someone lacking knowledge would just set the level to +8 (and the corresponding -4dB reference level on the Lavry--as it's clear the numbers have to add up to 12 to just avoid clipping the Lavry above 0dB).

However, I've long known about inter-sample-overs.  In between samples, the signal may rise above the 0dB level when rendered with oversampling--which must fill in the points between the points.   My sampler is likely to read these inter-sample-overs at least some of the time (or maybe nearly all of the time if I'm sampling at a higher rate than the original, which I need to do for decoding HDCD's for example).

I've long assumed that inter-sample-overs could reach as high as 6dB, though I've only ever actually measured about 4dB.  If in fact inter-sample-overs could get that high, I'd have to set the Emotiva gain no larger than +2 and the Lavry at -10 to avoid having them clip the Emotiva.  (But read on...)

Now, if you're not understanding inter-sample-overs and why I seem to be setting the MOL from the Emotiva at 6dB lower than the optimal distortion level (+8) for no good reason, read Benchmark's description of inter-sample-overs.

Benchmark sets their headroom for inter-sample-overs at 3.5dB.  Perhaps I'd measured 4dB because of rounding in the low resolution readouts on the I've been using.  Benchmark says the  maximum theoretical seems inter-sample-over is 3.01dB around 11kHz.

If only 3.5dB headroom for inter-sample-overs is needed, since I can't adjust in fractional units, I'd have to allow 4dB headroom below the optimal distortion at +8, thereby putting me right back to the adjustments I was assuming in my first "optimization," +4 gain on the Emotiva and -8 "reference level" (ie 8dB gain) on the Lavry.

So that now does look like the optimal adjustment, taking both headroom and noise considerations into account.  No only will no real signal cause rising distortion from the Emotiva, but no inter-sample-over will either (and many converters don't even handle those well...perhaps the true deficit in CD reproduction from the beginning).  In fact with the +8 setting I've allowed at least 0.9dB more headroom than necessary for inter-sample-overs, and possibly more (as I never bothered to measure the +9dB setting on the Emotiva).

(Note that the Oppo BDP-205, which was playing but paused during the noise measurements, and played the 880 Hz maximum level signal, has 2V XLR output.  Some say this is "all that's needed for any amplifier" but in fact my Krell FPB-300 required 2.8V and much higher to reach the true peak levels.  Can't always rely on inter-sample-overs to bring you there either--those depend on high frequencies usually.  I would have expected standard XLR level to be twice the RCA level, so 4V balanced. Anyway, with the Emotiva at 0dB it will also be putting out the 0dB level of 2V which is pitiful I think for a balanced output.  By boosting that 4dB, it's being boosted to 3.2V, and allowing another 3.5dB of headroom would put us around 4.7V.  Surely the Emotiva balanced outputs are still in their low distortion range at that point!  In fact, in the Secrets of Home Theatre and High Fidelity measured the lowest distortion at 5V !  (It bothers me they didn't probe the question "how high does it go", but as it turns out, 5V is all I need.  The 8dB gain setting on the Lavry officially corresponds to a voltage of 16dBu or 4.89V.)

Now I sort of remember that I'd come up with this setting (at least the -8 reference level) many years ago, driven by the Two Against Nature or Everything Must Go, both Steely Dan albums had incredible inter-sample-overs.  I figured out this same setting (and I think it was this one) empirically driven by the need to play the a Steely Dan DVD-Audio, because when I first played it, it clipped my sampler like hell.  This was how I personally discovered inter-sample-overs before I even read about them.  Benchmark refers to a particular Steely Dan track on Two Against Nature in their discussion and then analyzes a few others on that album and a few other albums notorious for the highest inter-sample-overs.

Then over time, I couldn't remember my results, and I drifted back to the obvious (-11).

Update May 12 2023

Oops, I wasn't thinking clearly.  While the "-8" reference level (ie 8dB gain) on the Lavry is still correct (that means the Lavry clips before the Emotive begins to distort...as things should be) the corresponding +4 gain is ONLY correct if there are no Intersample Overs (ISOs).  Only if I played a recording without ISOs, like my 880 Hz 0dB test disk, would that be OK.  Albums with a lot of ISOs would need to be played as low as +0.5 on the Emotiva, if Benchmark's claim that the maximum ISO is about 3.1dB.

I got plenty of ISO clipping when I tried to copy one of my all time favorite DVD audio discs, Pulse, by The New Music Consort.  It clipped with the Emotiva set to +4dB gain, clipped at +3dB, and so on.  Actually, it was still clipping at +0dB, which suggests maybe even Benchmark was wrong, ISOs may occur even greater than 4dB.  In this case, I was recording a 24/96 disc at 24/96.  I would think that would minimize ISO but maybe not.

The clipping at 0dB made me think that maybe my filter selection on the Oppo was in not optimal so I checked that.  I was thinking perhaps I was using one of those audiophile 'slow' settings.  But I found that I was using Linear Phase Fast.  That should be fast, in fact I'd think it was the fastest and best so that's why I chose it.

I looked over all the settings again.  I wouldn't want any of the 'slow' settings since those always leak aliases.  I looked over Archimago's investigation of the different filters.  IMO the only settings worth considering are Brickwall, Minimum Phase Fast, and Linear Phase Fast.  The Apodizing filter is weird having rippled HF response.  Minimum Phase Fast is Oppo's default, and it puts the ringing entirely after a pulse and the ringing is quite long, I consider that weird too.  The 'Corrected Minimum Phase Fast' is actually a slow filter having similar high frequency cutoff as the slow filters.  My subjective judgment of all the graphs and statistics is that the Brickwall filter is best.  It has the lowest distortion and noise, and second to the flattest response at 20kHz (according to the Rightmark evaluation) with the Minimum Phase Fast filter being 0.01dB flatter at 20khz, hardly worth sacrificing any noise or distortion for.

Audiophiles are inclined to dismiss brickwall filters I suspect mostly if not entirely by suspicion.  But this is not your grandfather's brickwall filter, this is a very precise brickwall filter.  (Archimago has no actual test of high frequency phase, which might show some way it could be inferior.  But the response is so flat and extended, phase errors must be pushed out in very high frequencies too.  And this is probably achieved without using any of the non causal FIR tricks that the linear phase filters use.)

Anyway, I tried it, and it might just be chance (caused by dithering or something) but it had fewer clipping events than the Linear Phase Fast filter.  With the Brickwall filter I simply had to repair one clipping ISO and there was almost 0.5dB of additional headroom left.  For the Linear Phase Fast there were 5 clipping events having to be repaired, including the weird wider than usual one shown below.

Brickwall (top) vs Linear Phase Fast

The Brickwall doesn't clip with this apparent digital error on the recording shown above, whereas the Linear Phase Fast does cause clipping (above +4dB headroom allowed for ISOs).  And the Brickwall ringing is slightly more compacted and seems to make more sense.  Generally, I liked the way the ringing on the ISO's looked with the Brickwall, in addition to having fewer of them clip.  I think I'm going back to Brickwall (it had been my option until a few months ago).  My feeling is that the Brickwall is less tricky and more honest than the other filters.  Among the filters, Brickwall may well be the mathematically least complicated.  Many of the others HAVE to be implemented with FIR digital filters because they are acausal.  They are approximations of things that are impossible to achieve with ordinary circuits.  Brickwall can in principle be implemented with IIR filters (but the Oppo may well use a FIR to approximate a more perfect Brickwall).

Indeed all the ISO clipping events on this recording are caused by ringing on very rare and unusual short impulses which look like digital errors.  Filtering out those pushes the peak level down to -1.8dB.  But even THAT peak level is still being caused by ISOs.  With no ISO's, the level should be -4dB, allowing my LF sinewave derived level setting of +4dB gain.  Here is what another of the ISOs looked like with the Linear Phase Fast filter.  Note that it appears to stem from a transient less than 4 samples wide at 96kHz (actually the impulse itself is less than 2 samples wide, or about 48 kHz, and therefore unlikely to be of acoustical origin).  Repairing some of these peaks may make them MORE audible as the inaudible high frequencies are replaced with audible lower frequencies which probably shouldn't be there either.  All the better to have fewer ISOs to repair, as with the Brickwall filter.

I made the last Pulse recording with the TV that I had used to set the filter choice still running.  To see if that made any difference, when the recording had completed one more pass (it keeps repeating endlessly on these Classic Audio DAD's) I stopped it, and kept recording for a minute, then turned off the TV, and tried to run for another minute (but it was shortened to 40 secs because the audio file had reached maximum length).  Amplifying the noise by 94dB, the point at which the TV was turned off is simply not visible.

TV turned off midway here makes no difference

That noise level, around -99dB peak unweighted*** (so approximately -120dB A weighted) is way lower than on the recording itself.  When I amplified the initial part of the recording, including before the Oppo started playing the disc, you can clearly see (at 36dB amplification) where the disc starts playing, the noise just rises out of nothing (the earlier -97dB peak unweighted noise from my chain of equipment is invisible at this level of amplification):

*** The actual value of noise measured may depend on how long you measure, when you measure peak noise.  There is sooner or later always a higher peak.  This is especially true given the ultra low frequency noise found in the Emotiva left channel...it is the primary driver for this noise dispersion, despite being at a tiny -103dB peak level itself.  In one second that low frequency level has barely changed, but in one minute it can do a lot of up and down dancing.

So the numbers I'm seeing now from the Emotiva at +0, the Lavry at -8, taken from the interval shown in earlier picture before where the TV was turned off, are these

1 second        -100.5dB

10 seconds     -98.7dB

1 minute         -97.7dB

2 minutes       -96.7dB

An advertising dept might see how small of an interval they could possibly go...  But nobody uses peak noise for specifications anyway, they use average noise weighted, but I can't do that as easily with Audacity.

I might establish a "10 second standard" for measuring this, though brag about the 1 second level.

(*In sighted listening tests.)

(** I know that DVD-Audio discs can be played through HDMI which preserves the full original 24 bit resolution, though it's still not possible for either HDCD or SACD on my system.  But it's still way better to have PCM computer files on my hard drive for automated playlist playback, and no files can be created from HDMI because it inhibits digital copying.  Unlike some, I believe 24/96 digital encoding and decoding is essentially perfect for any audio purposes, and the present results support this by showing digital conversion to be good enough to show flaws in an excellent preamplifier.  I see no reason to store DSD files as they would later need to be converted to PCM anyway for playback through my digital crossovers and equalizers.  So I might as well do the conversion (SACD, HDCD, or DVD-Audio to analog to digital 24/96) before storage, and it saves me a lot of hassle later too, when I'd be using the exact same components to produce 24/96 for my crossovers and equalizers anyway. )

The Button

It's been at least a month since I finally implemented The Button (the Listening Position EQ button).

I keep The Button next to the listening position chair.  When I sit down in the chair, I can press The Button and it enables the special Listening Position EQ which has much too much deep bass content to be used when I'm playing the system as background music (which is what I do most of the time).

So, when you press the button, you start getting the Full Effect at the listening position, which otherwise sounds very bass shy compared to the rest of the room (and the entire house).

Last year as I was doing some listening position EQ adjustments by ear and measurements, I decided that a Listening Position EQ was absolutely necessary to restore the missing "punch" in my system.  One problem is that there most of the notching out of room modes benefits the sound in the rest of the room.  At the listening position, these modes cause cancellation instead of augmentation.  So two EQ's are necessary if you use the system for both background and serious listening, especially for my problematic room set up.

(Why didn't I fix this by "fixing the listening position and speaker location(s)"?  Because my living room is decidedly a multipurpose room I also use for parties, and the only real solution to getting the bass right in a small lengthwise room like mine is to put the speakers in the middle of the room and listen at the back, as suggested by local late audiophile and engineer Alan--and what he did in his dedicated upstairs listening room, which had similar dimensions.  Putting the speakers in the middle of the room just wouldn't work for me.)

So last year I created two different EQ's.  "Listen" for the Listening Position and "Back" for Background.  I could fairly easily select them with the chairside EQ I was using for level and EQ adjustments, using stored memories.

But in December, the particular EQ unit that I was using as my chair side EQ died.  It is still on my bench awaiting repair.  The key stumbling block is that I have to deal with lead-free solder for the first time, and to do it right I also have a new powered solder sucker that may take some getting used to.  So while there has been one urgent thing to do after another here at the palace, otherwise known as Peterson Studios and Laboratories, that particular project of repairing the EQ has been procrastinated on time and again.  Which is not unlike many such things in the past.

Meanwhile I was forced to recall, guess, and scrape my blog for hints as how to recreate the "listening position EQ" using my Bass EQ unit, since sadly it was one photograph I never took before the chair side EQ died.  I did that in January.

But the truth is, I wanted to have The Button even when I also had the Chairside EQ working, simply as I believed it would be more convenient to use the button than fiddle with the memory menu on the EQ.

In one fantasy, I'd just sit down in the listening chair and The Listening EQ would automatically engage, then disengage when I got up.  That *would* be nice.  But even just having a simply "On/Off" button to press would be pretty close.

I finally got to it during a particular stretch of wintry weather.  The basic technology is stuff I'd used before, but had to refamiliarize myself with.  

I used an Midi Solutions Footswitch Controller (which I originally purchased many years ago for controlling a Behringer DCX crossover unit to change the absolute polarity, for absolute polarity blind testing.  It was controlled by the line level path of my ABX switcher).  I programmed the Footswitch Controller to send Program Change 2 when the circuit is made, and Program Change 1 when the circuit is broken.  The Midi Solutions app strangely doesn't code this from any menu option.  The only menu option available is to send a Program Change when contact is made, but nothing when contact is broken.

Meanwhile, the "circuit" is made and broken using an Insteon IOlink, same as I used for other automation tasks in my living room stereo.  (This is the 4th IOlink in use for the Living Room stereo).  The IOlink conveniently has connectors either for turning an internal voltage on or off, or making or breaking an external connection.

So then the IOlink is controlled by my Insteon system, which itself is controlled by my Universal Devices ISY994i home controller.  The Button is linked to control the IOlink.

It all works, but apparently (and strangely) the Insteon RF is a bit weak at the listening position, so sometimes I have to press the button more than once.  It's obvious when the Listening Position EQ is turned on, because a message temporarily pops up on the DEQ screen, the Memory light turns on, and also a light on the IOlink lights up (I could turn that light off, but I've decided to keep it as an additional indicator light), because if I were fiddling with the memories beforehand the Memory Light might not correctly indicate the Listening Position EQ, whereas the light on the IOlink is unambiguous.

I also have home control system automatically return the EQ to Background EQ when I press the Bedtime button at night, which also turns off the stereo.  So in the morning when I turn on the background music (the local FM classical station KPAC) from the bedroom it doesn't start blasting bass because I had been using Listening Position EQ the previous night but never turned it off.  This is important.  Automatic resets are important when an automated system is restarted.

[I plan to add pictures to this post.  But lack of pictures was what delayed me from posting about it for over a month, so I decided to proceed without them at first, which has been my usual approach anyway.\

Playing "Copy Protected" CD's

I've been trying to copy an old CD in my collection (Dave Grusin and the NY-LA Dream Band) on to my harddrive.  The disc itself was simply not recognized as a disc by my Mac and I had to disconnect the DVD-R drive to get it ejected.

Then I tried making a copy using my SmartAndFriendly CD copier.  That seemed to work (and it said PA meaning Passed) but once again, the computer wouldn't load it.

So then I tried playing it on my Denon 9000 (since I was streaming through the 205 at the time).  It showed the track counts but never got past 0:00.  It positively locked up the Oppo BDP-205, which I had to shut down, and then press "eject" to start the player up before it would try to load the disc.  The Oppo BDP-95 locked up similarly.

But I popped the disc into the Pioneer PD-75, and it started right up.

*****

There was a fairly brief moment in the mid 90's when record companies decided to thwart the newly possible copying of CD's (now that computer CDROM drives were available).

They threw in bogus crap in optical surface or digital stream so that the erstwhile CD's were not truly Redbook conforming.  But old style CD players that were not computers generally wouldn't notice the difference. 

It didn't work because it infamously didn't even work in all CD players, which themselves were becoming more and more like computers (a transition that was basically complete by the time you had DVD/CD players...which are basically little AV computers).

Some manufacturers were hoping "dual laser" systems would solve the problem.  CD's would get the "old fashioned" treatment.  But that wasn't possible because the resulting "dual laser" systems wouldn't read CD-R's (remember the Sony DVD-7000, a "reference" DVD player for at least a couple years, played CD's but not CD-R's).  Plus they were expensive.

Ultimately the audio industry bought up the recording industry and solved the problem for music simply by not giving a damn anymore about whether music copying could be technically prevented, but having million dollar litigation against copyright violators.  And also paying most musicians next to nothing for "streaming" music while just a few top artists make gazillions from multi modal appearances.

But that "not giving a damn" part has remained somewhat variable.  Copying from SACD's in the original DSD form was not possible for quite a few years until it was, but still not supported on most equipment.

(I see the SACD format as an attempt to bring back copy protection for audio, that's what David Rich concluded in a technical analysis published in Stereophile (!) in 2001.)

Anyway, nowadays many if not most audiophiles (like me) copy their own discs and other files onto their computer hard drives and use some sort of of software based system to queue up and play their music.  This is so much nicer than the olde days!

So then what about those "copy protected" CD's ?  Well, for this purpose, it may be necessary to have an old fashioned CD player which is not like a computer drive.

I have two such players (a Sony 507 ESD and a Pioneer PD-75) but only the Pioneer is close to hand.  In fact I set it atop the "for sale equipment" pile next to my system (only temporary, I said) even though it's not (anymore) for sale.  It's there because I didn't want to hoist it back onto the top shelf after getting it down thinking I'd sell it.

I thought it sounded special sometimes and might be keeping for that reason, I rationalized to my self.  Plus it's one of the cooler pieces of CD playing engineering ever, along with it's even more monsterous brother, the PD-95.  A friend was selling CD "dampers" for a decade.  I think it's ridiculous that such a tweak would help, but the Pioneer was doing the job the only correct way, by clamping the disc to a relatively heavy and damped platter.  So if having a stable platter was a big deal (and if you've ever seen a CD spin in a typical player, buzzing and all, you might think so) this was THE way to do it.  But after that, everyone just relied on digital processing to clean things up, which we now take for granted.  My friend positively hated the Pioneer and said it wasn't as good as his dampers on a Sony player (his favorite player from then and decades later was the Sony 990, a cheaply built player he started recommending to me as an "essential" upgrade only months after I'd purchased my first sort of "high end" CD player, the Sony 507 ESD, which had been his recommendation for the preceding 2 years...I still have the 507 and modified the drawer mechanism to eliminate the pesky "drawer sticking" problem...I consider the 507 my reference "early" CD player...it uses dual 1541A's in 8x oversampling, what converter you get in the 990 depends on which 990, either early AD or sony's own 1990 generation, which curiously enough was probably produced not only in the city I've lived in since 1992, but on the opposite side of the freeway from where I worked for 23 years, at least until Sony shut down their US semiconductor operations around 2001...I remember them shutting down and leaving the lot vacant for many years...that was when all the unique Sony made 1-bit converters used in the likes of the SCD-1 disappeared, and even Sony SACD players switched to using off-the-shelf multibit sigma delta converters from the likes of Analog Devices...while I believe those final 1-bits are the most special (as in "magical", not necessarily "accurate") of the lot for DSD/64 because it's native 1bit with 10x oversampling...1 bit "all the way" so to speak...which is why I hold on to my currently inoperative regularly laser eating DVP-9000ES as the poor man's version of the SCD-1...possibly useful for the ultimate 24/96 transcriptions of my SACD's using that uniquely good--or bad--approach).

(My feeling is that Sony ultimately concluded the full on 1-bit systems were just too expensive because of the necessary power supply heft to get them to work right.  The 9000ES was the least heavy ES player to ever use this technology, and it weighed 30 pounds with copper chassis and everything.  This was true of the previous decade of top end Sony CD players as well.  IOW, the true 1-bit technology just was not scalable, in addition to having a bunch of curious very low level problems at high frequencies (idle tones) which are probably not important or perhaps even euphonic....perhaps best to leave those in there just as with the ripples in linear phase PCM decoding.  Then you get the full magic, just as when SCD-1 was introduced.  But also it was just deciding their separate US semiconductor operation simply cost Sony too much, and it already looked like they weren't going to rule the world with 1 bit audio, though they just might do so for high resolution video...which was being done elsewhere and they wanted to focus on that.  Perhaps those business considerations were even the only reason.  Or, perhaps multibit DSD/64 reproduction was better, which seems to be what everyone but me thinks.  Though PS Audio has sold converters derived from the original Sony approach for some time, so I'm not the only one thinking it might be special.)

*****

So now the PD-75 has an actual "purpose" though I think it hardly justifies having such a valuable player "just" for copying a few CD's onto my harddrive after conversion to analog via the PD-75 and back to digital on the DA-3000.  So far only one CD has needed this special treatment.

My earlier rationalization was merely that it might make some discs sound better, presumably because of some euphonic alteration (such as peak unlimiting...Pioneer is a nicely oversampled player with good enough electronics to do the high oversampling it does..."Legato Link").

If there is such a euphonic effect, I believe I can capture it sufficiently (just like anything else) at 24/96.

In sighted testing, I always feel the Pioneer is special.  And it also makes less mechanical noise than just about any player.  That may be the key benefit.  Probably players like Esoteric are quieter still (and Estoric had a mechanism that was similar in concept to the PD-75 but even more elaborate and heavy).

1000 Albums

Today I hit a minor bragging point milestone, I have now copied 1000 of my audio albums (that I have purchased) to my hard drive for playback with Roon.

 I wasn't thinking this was a lot, but some might.  I know audiophiles with 4000 or more albums on their hard drives.  And lots of audiophiles, me included, have over 1000 commercial albums in some combination of formats such as LP's, CD's, tapes, MP3 files, etc.  But it takes some work even to get CD's copied to the hard drive correctly   (though I now have a fairly streamlined and decent process using XLD), with the metadata correct, in the best folder, and checked for errors and avoiding duplication.  (I'm figuring 20 minutes median average time when there are no issues, but sometimes it can literally take days of cleaning and stuff to get a good copy.  This is especially true of digital copies of vinyl albums.  But even plain old CD's can be a lot of work.  Recently I tried to get around the strong buzz made when I was reading a particular CD.  So I copied at 4x onto a CD-R using my SmartAndFriendly CD copier.  Bu when I read that copy onto my computer with XLD, it had lots of Accurip errors, only one track was copied correctly.  So I went back to the original CD and copied it using my computer DVD+R drive.  Even with the loud buzz it copied correctly.  I'm now a bit worried about the SmartAndFriendly copier which I thought was perfect.  This isn't the first time I've made an "inter-copy" and in previous cases it was the best solution.  To make sure my backup disc wasn't harmed in this process, I removed the backup hard drive from on top of the DVD+R drive, after turning off backup and shutting it down.  Etc.  But usually it's the metadata which causes the most grief with CD's.)

Large numbers of audiophiles have access to millions of albums online through services such as Tidal, QOBuz, Apple, Google, Youtube, and others.

But somehow "my" albums are much more important.  If I go randomly searching through catalogues such as Tidal, I may have to listen to quite a few albums before I find one I like, or even tolerate, even in categories I think are good.

Sadly, this might even be true of a few "my" albums.

And this got me thinking of back in the early days of my audio hobby.

It has seemed to me like other people always knew exactly what albums to buy, and which to play, but this was never true for me.  I struggled at record stores.  How did anyone ever know what to buy?  More than once I had a friend pick out good albums for me at a record store.

Now my tastes were broad, but also very peculiar.  Never much listening to AM radio (which was BIG when I was a kid) and avoiding commercial rock and pop stations on FM, I was not tuned-in to any stream of recommendations like cool kids were.

I mostly liked classical music, always touted by my mother.  Though it seemed like most of our records were show tune albums, beloved by my older sister.  I only started listening to the radio in 7th grade (after buying a clock radio, I quickly progressed audiophile fashion to having my growing "hifi" turned on by a timer) but didn't do much until 1969 or so, when I bought a reel to reel tape recorder from a friend who was leaving state.

THAT was when I really got into listening to FM radio, at least for awhile.  It always helps to have "something to do."  But the radio I listened to was almost entirely non-commercial stations like KPFK and KCRW.  So I still wasn't getting tuned into ongoing popular music (or even "classical" music).  The tape recorder had a foot pedal to turn pause on and off.  So I'd have it set up in "record pause" and press the footpedal, by the side of my bed, when I heard something really good, and I created tapes of the best "songs" and "episodes."  But as would be true with VCR's and DVR's decades later, it's very likely I spent more time lovingly creating these tapes, than actually listening to them.

I only learned of the Beatles because that same friend who sold me the reel to reel tape recorder, and then he was gone, and my next friend was more into Crosby, Stills, Nash, and Young, which I didn't much care for.

So I really didn't get connected to popular music in high school.  Only classical music, which I was performing as a student cellist in the High School Orchestra.  And whatever I happened to hear otherwise casually.

But I had two different sets of friends introduce me to Pink Floyd when I was in College.  I heard Dark Side of the Moon in December 1973 while getting high on marijuana for the first time in a friend's dorm room.  Later some really cool girl invited me to a party when Wish You Were Here was being rolled out.

In the late 70's one friend of mine really loved "The Wall."  And yet another got me into Steely Dan.  A very nice (but way out) guy I only saw a few times got me into Eno.  Also some were into Progressive and Electronic Rock (King Crimson, ELO).

So that's about it, about all I knew, in popular music, by the time I was 24 (1980).  You couldn't just look up endless articles about any band online like you can today.  You either kept up with radio, relevant magazines, or had lots of cool friends.

I met an elderly Jazz Musician in the 1980's, and he helped me pick out some albums.  And some girl I danced with did likewise.  Finally I was liking New Age Jazz generally, which featured prominently on a San Diego radio station.  I might have bought a Windham Hill record or two, but somehow it wasn't as catchy as the radio.  (Only about 10 years ago, I rediscovered what I'd been missing all along.  Shadowfax and Pat Metheny.)

Now I know a lot more, obviously.  But still it's hard to find new things, and I simply hate all the recommendations given to me by streaming services.  They don't know my taste at all, or actually it seems like they know exactly what I like and then insist on giving me the exact opposite.

Firstly, for more of the time, I'm looking for music without words, and that's virtually impossible to find on streaming services.  There is No Such Category, since any kind of music can come with words, especially when you venture beyond known classical composers.  I myself create Electronic Music (including my latest album, Mythic Rocks, now on all streaming services) with no words, but if I try the category of Electronic Music on any particular streaming service, nearly every song recommended will come with words...  I've even come up with several conspiracy theories to explain this.  They need the words for subliminal brainwashing of some kind.

Anyway, that's one reason why it's good to have your own personal collection of music, though nowadays you could just have use the "playlist" or "favorites" or "My Album" features of whatever streaming service(s) or music manager(s) you are using, and accomplish the vital "pre-selection" of music without actually "owning" anything.

But then there may be things that are hard or impossible to do.  Right now I mostly listen using my mplay playlist generator (which now features splay and tplay for even better play management) and it's easy to make playlists for music on my hard drive, but not so easy to make textfile playlists of albums on streaming services...if it's even possible I haven't yet figured out how to do it and they might never make it possible).  So right now I need lots of "owned" music to keep my all-waking-hours automatic jukebox working.

And there are other things.  Often there are individual tracks which are duds and I really want to remove them.  That's much easier to do (or even uniquely possible again) when you own the music files.

HDCD recordings need to be properly decoded, which is not possible on my system (or most nowadays) without doing it in advance on the files.

Finally, I have even gone to the point of editing tracks of music when they have peculiar misfeatures but are otherwise indispensible.

But the "automated" (scripted) playlist generation is the big thing for me now.  I'd get much more use from the streaming services I use if they let me create and use text playlists.  As it is, I'm thinking of cancelling Tidal and keeping just one paid service, QOBuz.

Anyway, back in the day when I was struggling all by myself to pick out new records at the record store I'd end up with a variety of conundrums.

In generally they'd be pushing "the latest thing."  But "the latest" album of some band you knew little about might well be their worst.  What you'd really like would be the "best" album by that band.  To find that, you'd have to get away from the end-caps and other displays and into the long aisles of recordings.  Even then you'd have a hard time figuring out which album of some band would be the best one to buy.  If you asked the sales person, likely it would be "the latest" which would be recommended.

I loved reading the recommended lists in The Absolute Sound and Stereophile.  But I could never remember what had been recommended when I got to the record store.  (And at most record stores such things would be impossible to find anyway...).

Somehow there was always a huge mental gap between the "independent" recommendation of albums and the purchasing process.  And even in our hypercommercialized online world, it seems there still is.

Then back in the Vinyl days, whatever I bought, after some long protracted selection process, would invariable be warped or have some skip or something.  And I'd never get back to the store on time to exchange.  Or if I did, the next copy (preselected by the store staff) would have an even worse problem.

So it's not surprising that when I get down to the never-before-copied parts of my collection, I could be running into a lot of "clunkers" I really don't want to listen to that much.

OTOH, it may just as well be, and actually is most of the time, more like re-discovery.  Things that sounded awful in days back when they were put at the bottom of the pile are now sounding amazingly good.  At various times in the past, my system of the day could make some things, and perhaps a lot of things, sound awful.

I didn't myself get a decent handle on this until I started doing acoustical measurements, around 1998.

But even then, there were endless issues, including amplifier bias problems, and even the discovery of the requirement for a down tilted response on electrostatic systems which was only my discovery last year, when I applied a slight boost at 1kHz to make everything fit this curve better (and sound much nicer).  I've made the most progress in the actual "sound" since I've been retired, now nearly 4 years.

Anyway, gems are turning up as I get to the bottom of my bedroom rack.  Such as Laura Marie, a musician I heard at a San Antonio Luminaria.

There's still much work to do on this pile, and there are many other piles.

One thing that this copying work does is create a new opportunity to listen to things as if they were new, again.

And I also benefit from my mplay playlists which ensure I ultimately hear everything.

In addition to decoding (or transcribing...which might be slightly better in some ways) HDCD's, I try to make transcriptions of all DVD-Audios and SACD's  too.  Since my system runs on 24/96 and 24/48 anyway, there is no benefit from preserving DSD from SACD's or other sources.  (I personally don't believe there is any benefit anyway, all the important characteristics of DSD are captured by 24/96, nothing better than 24/96 is needed for anything except possibly production.)  And vinyl too.  I prefer to transcribe vinyl whenever I'm playing it at all.  And those transcriptions become part of my collection too.  Perhaps even multiple transcriptions, comparing different players, turntables, or techniques.  (I put some redundant copies in a different folder so they don't get auto-selected for playlists.)  Actually despite thinking about it a lot, I don't have that many LP transcriptions (and many I do have are ready to be upgraded with better recordings on better systems).

Trick for when Album Leveling doesn't work

 Level adjusting is very serious business.  Back when I first got Sonos in 2006 and tried "random play" the difference in levels made many things unlistenable, often because too loud.  (And yet, I'm a person that generally likes "realistic loudness."  But in a playlist of music, something that is too loud sticks out like a sore thumb.)  I believe that virtually all the inexplicable differences that audiophiles claim to hear due to tweaks that have no explanation that is reasonable to audio engineering are actually level mismatches in disguise, since we can hear differences as small as 0.1dB (though it's not easy).

Now I rely on Roon's "Album" Volume Leveling, which works great in most cases.

Sometimes, however, it doesn't.  Typically when I am playing back programs recorded from FM radio, for example, which are highly compressed.  Or my own compositions.  Roon fails to dial back the maximum level to make up for the high degree of compression or high level of deep bass/extreme highs.  It does no good if I make the recordings at -6dB, Room simply adds +6dB gain.  Depending on various factors, Roon either amplifies up to peak level of 0dB or it reduces below that level if it thinks it's going to sound too loud.  But sometimes if fails to reduce, or even increases the level when it's not needed.

Two commercial albums that exhibit the same problem are Beautiful Vol. VII by Andean Fusion and Respighi Ancient Airs and Dances Suites Nos. 1-3 by Rico Saccani.  The latter is actually what inspired the present experiment.  Roon album leveling increases the level by 2.2dB.  It sounds much better without that boost.  Beautiful is already recorded at maximum peak level and Roon reduces it by a whopping 4.4dB but that's still not enough.  It doesn't help when in addition to being highly compressed, albums are very dry (little reverb or ambience).  This can make recordings sound painful even in the absence of distortion or clipping.  "In Your Face."  A slight distance is more comfortable.

I discovered I could stop this by appending a "needle drop" click (excerpted from recording vinyl) amplified to maximum level.  But needle drops at maximum level are a bit annoying.

A single cycle at 1kHz was just as bad or worse.  I didn't even want to try a single peak cycle at 20kHz, it would exercise the maximum peak level on a 150W amplifier that is powering my supertweeters.

But my subwoofers have internal digital EQ which steeply rolls off bass below the tolerable limits of the woofer.

So I tried a single peak cycle at 1 Hz (1 sec).  It works well, and I hear hardly any sound at all (just a very soft click).  This can be easily added to the end of the last track of an album, or any digital recording, using Audacity which can generate any tone, even 1 Hz, of any duration down to 1msec.  While playing a single peak cycle at 1 Hz, I see the digital meters on my EQ going to 0dB slowly and back.  With a peak level signal, Roon will not add any gain.  When the 1 Hz peaks at only 0.999 in the Audacity generator level control, Roon adds +0.1dB gain, so I decided might as well go all the way to 1.0 then Roon won't mess with it (Roon can't, other than lower the level).

(The first time I did this, I was scared out of my chair by a temporary shadow caused by a plane flying overhead just at that moment.  I didn't intend to invent the black hole speaker.)

I'm not sure if this would work on all systems.  Most do reject deep subsonics quite well however.

Reducing hum from -94dB to -100dB

It seems like I can improve the S/N from the Denon DVD-9000 dramatically, over 6db, with a simple tweak which might apply whenever unbalanced interconnection is used.  (Or it might be peculiar to this set of components, etc.)

I connected an extra set of stereo cables from my Denon DVD-9000 to the Emotiva XSP-1 Preamp.  I deliberately connected the unused surround outputs to an extra of the preamp.  As the outputs are not carrying any signal during this test, they may act as a better ground connection than the initial stereo pair of cables connected to the main output.  It also turned out that just adding one unused cable made nearly all the difference.  Adding a second unused cable gave little additional improvement, so I suspect more similar cables would only be for the very obsessed.

The results are shocking.  I got over 6dB improvement in peak unweighted S/N ratio from having the extra cables.  Most is clearly hum but some is noises that have a 60 Hz period.  This is similar to improvements I see with balanced interconnects over single ended.

For the test shown I had the Denon DVD-9000 playing a regular CD (not HDCD which might be tricky) but in pause for the entire duration of the test.   The Emotiva XSP-1 gain was set at 0.0dB and not changed.

In the starting condition, I had the two extra cables connected from the surround outputs on the Denon to another input on the Emotiva.  At two minutes, I removed the first extra cable on the Emotiva side.  At 4 minutes, I removed the second extra cable.  At 6 minutes I removed the one of the cable connections on the Denon Side,   Around 8 minutes and 40 seconds I removed the second cable.  Then about 10 minutes I reconnected one of the extra cables.  Around 11 minutes I reconnected the second extra cable, so it was back to the starting condition (to be sure nothing else had changed...and it hadn't changed appreciably, compared to the main effect, which is huge).

DVD-9000 with and without (center) extra interconnects

There is visibly far more hum and noise (it's mostly hum but with some strange high frequency bursts that repeat every 1/60 sec) in the center condition which is just using the two stereo interconnects.  As soon as just one unused extra connection is made, the hum and noise drops 6dB.  Then when the second cable is added it improves a few tenths of a dB more.  What is remaining with the extra cable(s) seems to be almost entirely random high frequency noise and the high frequency bursts at 1/60 sec are eliminated.  It seems hardly worth trying more than two (I have no more good cables to try anyway...I had to buy this pair last week)...but just adding one unused cable is huge.

(Possibly it has something to do with the grounding on the Denon DVD-9000...but everything about the DVD-9000 seems impeccable.  BTW, I have Direct Mode set to 2 which shuts off video output.  Also it could have to do with the grounding on the XSP-1.  The extra cables were connected to Input 3, which is in a different bank.)

Another curious thing is that even connecting the cables on the Denon side alone seems to help.  I also saw that kind of behavior with the Video output--having a stub video cable connected (but not connected to anything else) makes the DVD-9000  quieter, even in Direct Mode 2.

Something that might be even better is connecting a very heavy wire, possibly mostly buss bar, to the grounds of two plugs and use that as the "extra" interconnection.  I think I've seen things like that sold as tweaks before.

Both my original interconnects and the extra interconnects are both Blue Jeans Cable LC-1, which has the best single ended shield configuration and plugs for hum and noise reduction.  The originals, unabused, are just a few years older.  These are very solid and thick cables custom made by Beldon with a double braided shield, and terminated by Blue Jeans with very solid Canare RCA jacks.  It's hard to imagine they could be broken or even slightly out of spec.

(Somehow, leading up to these tests, I had intended to upgrade the previous cables to LC-1 to see if that helped.  From above, it looked like I had been using standard grade Radio Shack cables from the mid 2000's.  Possibly intended only as a temporary fix until I got better cables.  But when I actually got close enough to change the cables I could see they were actually LC-1's.  It's funny how when the picture is unclear my mind can instead give me a clear picture of something wrong.)

Previous measurements shows unweighted peak S/N ratio around 93dB for the Denon DVD-9000, and 95dB for the Oppo BDP-205.  That 2dB difference was entirely hum and showed the benefits of balanced interconnection.  Both players actually have "120dB S/N" specification, the limiting factor is my Tascam DA-3000, which can only sample with 113dB S/N, which like all factor specs is A-Weighted, meaning the unweighted S/N is likely in the 95-105dB range.

With these changes, it looks like the Denon will be either identical to or maybe even better than the Oppo, at least for practical purposes when sampling with the Tascam DA-3000 for transcription or playback through my DSP based system.

If the test seems strange...it's partly because it followed an earlier series of tests which were inconclusive because I could never be sure that either I had reset the starting conditions in an identical way, or that the Denon hadn't started muting in some extra way.  Leaving the player running the whole time and going back to the starting condition at the end were the ways to remove these uncertainties that arose in previous testing despite attempts to control the starting conditions rigorously.  The Audacity screenshot was made after 80dB of Audacity digital amplification.  The noise seems higher than some previous tests because the Emotiva volume control was set to 0dB instead of -4.5dB (which I had been using for HDCD since the Denon actually expands peaks at least that high above the normal CD-level 2V output, though I discovered on album Poems of Thunder that in some cases HDCD peaks reach 5dB).  So add 5dB to the wideband unweighted peak S/N levels for comparison relevant to HDCD which is mainly what I use the Denon for.

Here are the peak levels from various parts of the test (correcting for the 80dB gain but not the additional 5dB available output on HDCD).

First 2 minutes (extra cable pair connected)

-91.05dB   (would be -96.05dB including 5dB extended HDCD peaks...this is better than previous data for Oppo which was -95.3dB)

Second 2 minutes (only one extra cable connected to Emotiva but both on Denon)

-90.3dB

Third 2 minutes (two extra cables connected to Denon but nothing else)

-87dB

Fourth segment (only one extra cable connected to Denon)

-85.5dB

Fifth segment (no extra cables connected to Denon)

-84.4dB (not including some early and late peaks which might be due to connection in progress)

-84.3dB (including suspicious early and late peaks)

Sixth segment (one cable reattached between Denon and Emotiva)

-90.0dB

Seventh segment (both extra cables connected between Denon and Emotiva)

-90.6dB

I could immediately add 5dB to all these numbers for HDCD because HDCD peaks are 5dB higher on the Denon, so I have to turn volume down 5dB on the preamp.  (Although, then the noise of the Emotiva itself would be a larger factor than it is, so results are not completely predictable.)  Then these are all unweighted peak measurements, which would improve by 10-25 dB with RMS and A Weighting.  And I don't know how much of these measurements are proportionally hum and ultrasonics.   It would be complicated to do weighted and RMS measurements on these data (I tried but ran into a snag) and only useful for bragging rights anyway since I don't see how I'm make these improvements significantly better without engineering a fake "grounding bus" interconnect as I suggested above.  This present improvement is great and I'll look for it in future connections.  (Though perhaps I should do more investigations to be sure this is a real effect and not a fault of the old cables, for example.)

My guess is that these numbers are showing at least 108dB properly A weighted RMS S/N, but with a mean guess of around 116dB S/N RMS A-Weighted based on the idea that RMS and A-Weighting would improve things about 20dB...but that can't be true since it exceeds the specs of all equipment I'm using (except the Denon itself, at 120dB S/N).  In reality all the noise levels are somewhat additive, so say if all 3 units (Denon, XSP-1, and DA-3000) had 120dB S/N, the result would be about 110dB S/N.  But the Tascam DA-3000 (balanced in) is at 113dB and the Emotiva XSP-1 is at 115dB (unbalanced in to balanced out).

Though the second additional cable gave only a fraction of a dB additional benefit, the 60 Hz component is still not entirely gone, so there might be some further benefit from grounding bus, etc.

*****

I have decided not to do any further investigations on this for the time being.  It could be, for example, that the initial cables were faulty (though they're fairly new and very rugged Blue Jeans LC-1), so if I swapped the brand new "grounding" cables for the initial ones, I might then have the benefit of low hum with only one pair.

Or it could be how the "grounding cables" are routed differently, perhaps out of EMI fields.

Adding cables might not always help.  I'm not claiming that at all.  I think only that if you see a hum issue, such a thing might be worth investigating.

In this cause there was no audible "hum" issue that I ever noticed.  It's deep down in the noise floor, deeply below the ambient noise in the room.  At -90 to -100dB or so in the electrical signal, at max output around 100dB, it would be 0-10dB, or -24 to -4dB "A Weighted".  If the two initial cables were somehow defective, and they would both have to be defective for there to be hum, one would expect much higher hum level than that.

There could be a straightforward explanation such as that each cable lowers the ground resistance.  Except it doesn't seem to work so incrementally.  Instead, the first extra cable made nearly all the difference.

It really seems like there were additional ground currents between the player and preamp, and routing these additional currents lowered the hum induction that occurred in the interconnect.

But why are these ground currents choosing to go through the additional cable(s) ?

Adding one more cable would in theory not be adding that much more conductance.

Instead, it looks like there is something peculiar going on.  Perhaps the grounding of the surround jacks on the denon is closer to the power supply ground.  And because of that, the extra currents DO choose to go through the new cable(s) selectively, giving a reduction in induced hum.

The Denon 9000 is grounded and so is the Emotiva XSP-1 (IIRC...or it's connected to grounded gear).  Ground loops may be to some degree inevitable, and perhaps it's a wonder they were already so low.

I think therefore that this finding is almost certainly some kind of fluke as I've described, not a general rule.

But it certainly looks like, I have already stumbled upon a lucky optimum.  Any change is likely to make it worse.  Then, assuming micro degrees in cable routing are also a contributing factor, I may find it hard to make it this good again.

And to do this test in the fairly incontrovertible way I have (as opposed to separate tests, in which some extra variable might have changed) means more wear and tear on my equipment, including my beloved Denon DVD-9000, which is very long in the tooth.  (In fact it already started acting up, with the drawer not pulling all the way in once, perhaps in protest.  Now I see it may do that so it looks OK but doesn't register the incoming disc, showing only 0's on the display.)  

To do a proper test proving the initial cables were the problem, I'd have to do lots of cable plugging and re-plugging while it's in pause with a spinning disc.  Almost every time, with the very tight LC-1 Canare connectors, I have to push so hard it jostles the Denon slightly.

To do this test, to make the Denon about as good as the Oppo (which uses balanced connections!), I thought it was worth this, but just once, not for a series of follow up tests to PROVE the causal mechanism, which may not even be possible.

To reduce such wear on my equipment, I'm ending this test here, at what appears to be some kind of lucky optimum in hum reduction.  "If it ain't broke, don't fix it."  At  least for now.  Next time the Denon is removed I'll know to check the cables again.