Saturday, April 22, 2023

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. )