Philmore FM Separator

Last night I found an old FM separator, made by Viewsonics, for Cox Cable.

I immediately set up the whole thing (which is a bit complicated) and it worked.

FM was about midway between as good as with the Yamaha direct on the higher antenna, and what it was before.

But things sounded wrong, grainy.  When I off tuned it on the downside it quickly became very distorted.

Today the new Philmore FM Separator arrived.  It is not like the one in the catalog photo (which, IIRC, looks like my Viewsonics).  It is much larger, made in China in a very solid box, and has specs (<= 2.5dB within specified ranges).

[Updated at bottom.]

It works far better.  There is no "off" sound and the noise is about as good as the Yamaha on the other antenna.

But I also wanted to start observing these things, so for the first time ever, I hooked up a scope to the F-26.

The Kenwood vector scope I bought a few years ago and had resting atop the F-26 for years had insufficient gain and was worthless.  I had to use my Tektronics.

It's clear the alignment is off, and the auto-lock pulls the station off tune, which is aggravated  by the, yes, narrow band that is always selected.  I tried some local rock stations, and they too seemed to be pulled in only in Narrow, even on the outdoor antenna.

I have a mind to surgically remove the auto-lock and auto-narrow, leaving wide the only choice.

And perhaps improve the alignment.

All looks pretty good, though, if I just tune right above where the tune indicator says.  Then the auto lock doesn't engage, and that turns out to be the best spot too.  And the IF modulation seems very flat, until it hits the stops on really wide modulations, because it's stuck at narrow.

No obvious issue with the new antenna system though.  I remember that Cox had their own FM channels and didn't care if you got KPBS-FM.

The optimal separator for a scanner would not be TV_centric, and just divide the spectrum between 108Mhz and 120 Mhz (so no division below 88 Mhz).  I'll keep looking for that.  But this one seems OK.

The new arrangement which made all this possible now has a shelf reserved for the living room tuner, so I could put another tuner there while I'm working on the F-26 more easily.

Something cool running with no drift.  The 9500 might be a good choice.  And now the stronger antenna signal means there won't be so much of a noise issue with tuners having more primitive MPX.

[Later tests demonstrated the Philmore 9500 is adding a kind of hashy distortion to 88.3 Mhz.  So compared to indoor antenna, the noise is reduced, but an irritating edgy distortion takes it's place.  I first noticed this on the already edgy sounding Bedroom System I have been trying to make sound better.  But then I found, indeed, I can hear the same artifacts on my other systems, they are just far less noticeable.  So then I tried a UHF/VHF seperator, which simply divides the spectrum at 300 Mhz.  Well it seemed to do the same thing!!!  (Compared running the antenna feed straight through, using a double female F adapter.)  Finally, I tried a mere splitter, which causes 3dB reduction in signal to both outputs at all frequencies.  This one claims to be good to 1500 Hz and is fairly chunky.  That does not add any hashy distortion to the signal.  I can barely tell the difference between it and the mere adapter, and to the degree I do, I might prefer the splitter.  It would seem the FM separator could have a problem at 88.3 Mhz, because TV channel 6 reaches 88 Mhz...any reasonable filter has to be in steep decline by 88.3 if not 88.2.  But why the issue with the mere VHF/UHF separator then?  Possibly there are ripples created by these things way beyond the separation point.  For low fidelity communications, these are adequate, but for hifi FM stereo, phase ripples cause distortions and delays in the demodulated signal.  And/or possibly also there could be nonlinear effects.  I noticed in one separator I opened, it had ferrite filters in it.  So I don't know whether the problems are caused by linear or non-linear effects.  Either way, a better design, using better parts, might be able to do better.  But after many hours of scanning, I've found nothing better looking than the Philmore.  Designing something like this is beyond my demonstrated capabilities, though I've seen discussions of making similar DIY devices, so I think I could figure it out after some time.]

EMF blockers

I'm also evaluating EMF blockers.  I have two critical needs.  I have a active room mode canceling speaker now placed in a strategic hallway corner.  It reduces some low bass modes significantly, and especially reducing booming in the hallway and bedrooms.  But when I turn on the light in the guest bedroom, it makes a huge pop on the speaker.  The light is also a fan which might have something to do with it.  I bought an EMF reducing power strip that eliminates the problem.  The only other measure that was useful was an isolation transformer.  But the isolation transformer made a continuous humming not good for the bedrooms.  But I don't like the long cord on the EMF reducing strip and can't find one with shorter or no cord, though I have one like that somewhere.

Also when I flip the switch on the transformer based Tensor lamp near the turntable I also get a huge pop on my stereo if I am playing phono or any analog source.  To reduce that I plug the Tensor lamp into a separate strip which plugs into one of the 3 strips my two racks use.  It would be better on a separate circuit but that is very hard to arrange.  If I plug a "quiet lines" device (by Audioquest I think) it reduces the pop a bit more.  That was what I originally bought the latest EMF blocking strip for but thought I'd try it on the room mode cancelling speaker first and haven't yet tested on the lamp.

Meanwhile I also bought a Greenwave AC noise meter for $135.  This is similar to but better than my Audioquest Noise Sniffer because it has an unambiguous digital readout and there is no "sweep" control to confuse matters.  Already I've looked at a few circuits and things.  My big Panamax 1500 indeed reduces all the noise of all my line level components by about half or third as on other circuits in the house.  But meanwhile, that circuit itself is by 3x the noisiest in the house, and, IIRC, that was also true before anything was plugged into it.  It's a dedicated circuit the electricians installed a few years ago.  I'm thinking I should have them check it out.  Perhaps if the circuit were quieter, the panamax output would be quieter still.

I couldn't find any new reasonably priced plug-in EMF filter/blockers, except for Greenwave units, so I bought 2 of them for $30 each.  They are sold on the basis that you need to clean up your household power for personal heath, and they want you to buy dozens of them, for every circuit or even outlet in the house.

A company named Cleansweep does make some pass through filters but they are not small wallplug types but boxes that must be plugged into wall with a cord, and they all look a lot more expensive than $30.

New FM Antenna for Living Room

I hooked up the kitchen tuner to play via Sonos in the Living Room.  This is very convenient, I can tune the living room system, which I like for background music everywhere, at the kitchen table.

But also I notice the kitchen tuner, currently the Yamaha TX-1000, is much quieter than the F-26.  I'm bad at estimating these things, but it could be as much as 10dB.

The Kitchen tuner uses the best antenna in the house, at the peak of the roof, a Magnum Dynalab FM whip.

The F-26 uses a strategic inside antenna location I discovered one day, seemingly better than other locations inside on KPAC.

I have a second whip antenna, a different brand, about 6 feet further along the roof.  I intended that for a police scanner, which I bought this year but don't use much.

But you can get such a thing as an FM separator, which separates the FM band from the rest of the VHF/UHF signal.

This is similar to, but far less common than, VHF/UHF separators/combiners, of which I have 2 on hand (and one in use) which allow you to combine or separate antennas or inputs.

In principle, there is no loss, because there is no actual duplication involved.  One side gets the FM band, the other side gets everything else.

WIth an ordinary splitter, you get 3dB loss on each side.

I have a coax link between kitchen and living room installed by electricians which has been rarely used.  I mostly used it for digitial audio, but I thought I had issues with it and gave up.  The actual cable is RG-6.  RF was one of the possible original ideas.

So I'm going to separate FM through a link to the living room and see how well that does.  I also need to use an antenna isolator to prevent hum, I had a Mondial Magic Box but bought a second one before I found it.  They're not made anymore and in my test worked better than a newer one.

It's funny how hard these FM separators are to buy.  There's one made by Winegard and one by Philmore.  The Winegard is out of stock at Amazon and every other store I checked.  I was facing a similar fate with the Philmore but found one store that carried it, so I immediately purchased it.  I see on receipt it's supposed to have 1Ghz bandwidth, which is good because the local cops are mostly 850 Mhz, but there are some lower ones, so I wouldn't just want a VHF/UHF splitter I think.

Another Brilliant Discovery (Solution later)

Glorious internals of L-1000T

I've long had a serious problem with my most favorite and wonderful sounding Kenwood L-1000T, which was Kenwood's last top-of-the-line FM Supertuner, following their previously renowned L-02T and KT-917 among others.  After building this ultimate FM statement piece in the late 1980's, Kenwood never again tried to build an FM tuner to this level of design, engineering, and construction excellence (they did build more sensitive tuners according to some reports later (6040) and at cheaper prices).  It came out just as Kenwood's competitors Yamaha and Pioneer were introducing their own ultimate supertuners (the TX-2000 and F-93) at a somewhat lesser level of perfection, and Sansui had already perished after producing their last gasp supertuner (the Tu-D99X)--a great but widely unappreciated tuner overshadowed by their earlier Tu-X1.

Or two problems, actually.

Funny the L-1000T has never been as famous or collectible as it's predecessors made by Kenwood.  Only a few well known reviewers, like David A. Rich, and the unrelated anonymous David at FMTunerInfo have even bothered to write about it.  David Rich only had a chance to look at the schematic and said it had the potential to be one of the best tuners ever, except for the digital varactor front end (which simply cannot be as good as old fashioned analog front ends with air capacitor tuning).  In every other way, he said, this tuner was unique in doing everything the best way possible.  Meanwhile David put it among the best 3 tuners ever, the best 5 tuners ever, or something like that.

This lack of more widespread recognition might be related to problem(s) the reviewers failed to observe or describe.  One problem many people have noticed is that it gets hot, very hot, almost like a Class A amplifier.  The second problem that I haven't seen much widespread comment about is that it can seriously drift over the course of 24 or so hours.  It can drift so badly that despite the quartz servo locked oscillator and everything you still have to re-tune the digital tuning after 24 hours for the best reception.  (It could be that only my sample has the drift problem, but I doubt it.  A friend of mine has noticed a similar drift problem with a nearly contemporaneous Kenwood FM tuner, the KT-6040.)

I used to keep 2 memories for every station I liked to listen to, the first at the actual station frequencies like 88.3 Mhz, and the second set for use after 24 hours of thermal drift, like 88.325 Mhz.  The L-1000T lets you tune to any 25kHz increment, which turns out to be very useful for dealing with the drift.  Useful, but the tuner would be much better without any drift.  Most digital tuners don't seem to have any drift that I can detect.  I've never detected any drift in my Sony 730ES, the very tuner that James Bongiorno tested for Sony and told them they needed to fix the drift, which he believed they never did.  But then I don't have anything comparable to Bongiorno's legendary FM tuner distortion measuring rig.  Bongiorno said that before the drift kicked in, the 730ES had the lowest stereo distortion of any tuner he had measured, but not for long.

But unlike my 730ES, which seems to sound continuously good, I could easily tell my L-1000T was drifting in two different ways.  The first was in the sound...it began to sound like there was serious interference on every station that was tuned to the correct nominal frequency.  The second, which I didn't comprehend at first, was on the tuning indicator the L-1000T has.

Even I didn't notice at first that the L-1000T has the kind of tuning indicator I had always lusted for.  It has a tiny LED representation of an tracking RF spectrum analyzer, similar in concept to the big scope display as on a Sequerra Model One with Panoramic Spectrum Analyzer (Panalyzer).

L-1000T with centered tuning indicator (vertical bar on right)

When a station is correctly tuned, you see a vertical bar in the center of the tuning indicator whose height represents the signal strength.  (I took the above photo with twinlead FM antenna resting on a chair, so the vertical bar is short, especially with Wide IF.)  If the tuning is off to one side or another, the vertical bar is on one side or the other side, or there are two vertical bars.  Two vertical or more vertical bars could also occur if there were interfering station(s)--that is what a panalyzer is most praised for showing most clearly.  Only the cheap LED version has the issue that one station could get divided into two rows of LED's if the station is not correctly tuned.  On a true scope display or a higher resolution LED display you might still see one line but more slightly off-center.  The crude LED version on the Kenwood requires more interpretation.

Correctly tuned line in center

Station to right of center, need to tune higher

Station lower than center, need to tune lower

What happened when it was drifting, and there was interference in the sound, was that indicator showed the two-line display instead of just one line (but unlike above two lines, they were above the center, requiring me to tune 25kHz higher).  And when I did tune 25kHz higher, the display went back to one line in the center, proving that drift had occurred and I did in fact need to tune to 83.325 Mhz to receive 83.300 Mhz.

Now maybe all I needed to do was re-tune at the 12 hour point, and in the between-times the servo lock was taking care of everything.  But I was always worried that during the 12 hours of warmup drift, the tuning wasn't right at any of the other times either, but somewhere in-between just too close to show on the tuning indicator or as interference in the sound, just as added distortion that might not be obvious.

And I did sort of feel that after 4 or more hours operation the L-1000T wasn't sounding as good as the first hour.  But that wasn't as clear and obvious and I could have been imagining it.  What was obvious was the big drift that occurred 12 hours or more after being turned on.

One way or another, the drift bothered me horribly.

Meanwhile, I never much worried about the heat.  I no longer stacked anything atop the L-1000T.  I made sure there were a few inches of ventilation space above it also.  And I tried to remember to turn it off every night, so as to keep it from getting too hot (and also having to re-tune every station the next day).  I figured that was the best I could do about the heat and that was that.

Now when James Bongiorno reported (on the FMTuners blog) to Sony about the drift in the Sony 730ES, he added that good tuner designers like his pals at Marantz designing the 10B had this problem solved long ago.  He didn't say more but I had understood it often required pieces that thermally tracked differently made to cancel each other out.  THAT was the kind of serious engineering I might have to learn to solve the drift problem.  But I was finally determined to solve it one way or another.  So after finally swapping the L-1000T out of my Kitchen Tuner slot I put in on my test bench.  That move also required several hours of cleaning the bench which had been accumulating flooby dust like expired or nearly-expired batteries I hadn't had time to re-test.  I was happy to have my test bench back in service for another miracle cure as I have sometimes pulled off before. 

By the time I was moving the Kenwood to my newly-cleaned bench I was figuring...this might not be so bad, all I really need to do is figure out how to keep it from getting so hot.  Since the drift only occurs over hours while it is meanwhile getting very hot, the two are connected.  It's the completely excessive heat of the L-1000T which is making it drift.  Otherwise, it might not.

And that has turned out to be the case.  But how do I keep it from getting so hot???  And especially, to keep it from getting hot near the parts that are responsible for the drift.

It may be the power supply has deteriorated and I should be replacing the power supply capacitors.  The power supply might run cooler then.  But I have observed the hot operation from the first year I acquired it, and it didn't seem overly used then, and many other people have observed the hot operation going way back.  Even if I do sometime replace the power supply capacitors, there's more to it than that.

By it's very design, and pushing everything into the best mode of operation, and with a stiffly regulated power supply which also helps, this tuner consumes a lot of energy for a tuner.

Back in the days when it was common for tuners to consume this much energy because they used tubes, tuners had ventilated chassis.  Many of the tube tuners I have had had no box at all, just a short base of electronics which tubes plugged into.  My first was a Fisher FM-80 with no case whatever.  I put it on a shelf.

By the early 1970's, Kenwood and others were making transistorized tuners but still venting the back of the tuner chassis to prevent heat buildup.  Then, sometime in the mid 1970's, Kenwood removed the ventilation holes from their best tuners, making bigger boxes instead.  That trend continued up through the L-02T.

The cheap and middling tuners in the line didn't generate much heat anyway.  But the big dogs, like the 600T, did generate a lot of heat internally and often show internal heat deterioration in and around the power supply.  I found a thread of people complaining about heat build up in the famous KT-917.  One guy says he wouldn't go to get milk without shutting it off.

Sometime around the mid 70's, a group of engineers left Kenwood and starting making their own Accuphase tuners (originally called Kensonic), which are still among the best, and far better than their Kenwood contemporaries, by many accounts.  Now you might think this was a matter of pioneering thinking engineers feeling they want to try newer and better technology, we in the US believe in such things.  But while Kenwood was busy pioneering things such as Pulse Count Detectors (which they never did very well, though there is still controversy about this) Accuphase was using more traditional approaches, such as ratio detectors, that had been used with tube equipment.  But there was one difference, that's obvious right away, when you just look at an Accuphase T-100, and a Kenwood 600T.  The Accuphase has vent holes in the top, and the Kenwood does not!  So it's clear, at Kenwood the future Accuphase engineers were indeed feeling trapped in the box.  And the heat too.  Not to mention which, the whole pulse counting thing, as deployed by Kenwood, was another way to make a tuner less sensitive to heat induced drift.  So while Kenwood was pushing the envelope of technology to keep from having to use holes in the covers of their tuners, the original engineers left to make tuners the old fashioned way...with holes in the covers.

When Kenwood started making low flat digital tuners, heat wasn't so much a problem anymore, at least with the mainstream models that might just use a few chips and that was it.

Meanwhile, the late 80's L-1000T has many high power consuming chips.  Kenwood didn't stick it into their smallest box, they couldn't, but it's still a fairly flat box, with as much power consumption as a receiver playing bookshelf speakers.

Why?  It seems Kenwood above others was worried about pepsi-syndrome or something.  Keeping tuners in an unvented box keeps them cleaner longer, and generally better working as far as alignment and stuff, except for problems caused by heat.  For the hotter running top of the line tuners, they should have taken more steps to keep power supplies cooler, I think.  And if ever that was true for any Kenwood tuner, the L-1000T is the ultimate worst example.  Many have noted how hot it gets.  The previous L-02T and others had been in a much larger boxes.

So a possible solution might involve something like drilling holes in the case, such as the top cover...

I first tried something much simpler.  I took the top cover off.  There was no evidence of drift or anything, and the power supply heatsinks reached 130F while the front end box was 87F.

The side piece on the power supply side was turned upside down and screwed on the least possible, leaving a 1/4 gap all around top-to-bottom.  Under this condition, the front end box remained as cool as before, and there was no evidence of drift for over 24 hours.  Problem solved!

However the innermost of the two power supply heatsinks was 157F, considered unacceptable to me.

I took the side panel completely off, and heat sink temperature still rose to 153F in 6 hours.  I'm thinking the heat sink should rise no higher than 150F or about 60C, a common design maximum.

The ultimate solution is going to have to involve holes in the top cover, which I will have  a machinist make, and possibly holes in the bottom cover as well.

Meanwhile I may have nudged the basic oscillator voltage adjustment more to spec, but it's still off a bit.  It changes far more during warmup than is ultimately off-by, and there is no change in the X-Y oscilloscope display at all three bandwidths (Wide, Normal, and Narrow) during or after the warmup.

The narrow is visibly off to one side, but this is invariant of warmup.  On an antenna wire laying on a desk.

My belief is the front end side has to be heated up very badly to overcome the servo-system that keeps the tuning locked on.  Otherwise it stays locked on and you notice nothing of the underlying varactor voltage changes because the servo instantly compensates for them (which is what is happening as you make the front end oscillator adjustment, and I've seen the servo jump volts in an instant).

But if the temps rose above 120F on the front end side, which they might do in the fully sealed box left running a long time, it might happen.  That could push RF oscillator to Front End alignment way out of whack.  The voltage is already rising from 23.48 to 23.83 as the RF box temperature rises from 75F to 85F.  Extrapolating on this it could be seriously bad if the front end rose to 120F.

Or likewise with the crystal, I think it would have to be heated up way above 85F to be a problem.

So I believe by letting the cover out 1/4 inch the drift problem is solved.  But the overheating is not completely solved, the power supply gets too hot, and that requires holes in cover, the one thing that Kenwood avoided in tuners after 1975.

Maximalism

 Many audiophiles, perhaps even most, brag about something like minimalism.  They de-contextualize Keep it Simple Stupid (KISS) to mean whatever approach they are using that they think is simple.

But if you then suggested, perhaps, something even "simpler," such as, say, an all-in-one phono player, or a cactus needle, you'd surely get scolded.

The full KISS principle is qualified.  

Einstein is often alleged to having said "Everything should be made as simple as possible, but no simpler."  What Einstein actually said was:

“It can scarcely be denied that the supreme goal of all theory is to make the irreducible basic elements as simple and as few as possible without having to surrender the adequate representation of a single datum of experience.”

Not surrender a single datum???  In audio terms, that would mean we should not do without anything that might make the audio experience better to even the slightest degree!  This isn't minimalism, it's maximalism.  We do more until the last datum is resolved!

Ok, so you may say that the "more" I am referring to could be eliminating extra "unnecessary" things.  But who is to say what's necessary and what's not?  How about doing without the preamp and just plugging your moving coil cartridge into your power amp?  That's simpler, isn't it?

I have never fallen in line with the simple-audio-system meme.  I've cynically believed its "simply" to get you to buy more costly stuff, with the claim that you won't need as many other things.  But then it turns out, in the long run, you may want those other things back.  I've seen that.

Instead, I seek to make the whole system as good as I possibly can.  That could, in principle, mean eliminating things, but that's almost never possible to do while retaining the same high performance.  I basically never added unnecessary things in the first place, at least if I could help it.

Perhaps I could design and build new things that combine just the exact stages, features, and so on that I need, with nothing else.  Trouble is, I have neither the time nor patience nor even knowledge yet to take on such formidable tasks.  Instead, I generally work with things I can buy, ready made and reasonably prices, and connect them together to realize my system ideas.  There may be more circuits, boxes, etc, than I need, if I could create some dream product doing all the things I need, and nothing more.  But no such product(s) exist, and even if they did, they would be incredibly expensive, as we can tell from products that do exist and are almost there.

And so it is that I have 6 DSP boxes, one set of 3 to handle my phase linear steep crossovers, and another set of 3 to handle other EQ, limiting, and display functions--because the other type does those jobs better.  And 3 of each kind because I need one for each "way" of the system.

Now I believe there is a system sort of like this, though not as flexibly as I would like, and it costs around $20,000.  My 6 DSP boxes cost under $400 each, for a total of about $2400.

The system has 3 ways because no one driver can cover the full audible spectrum well.  At least none that I can afford...

Anyway, now I may be adding a new gizmo, a self described "Buffer", the Musical Fidelity X-10D Version 3, at least to the path originating from my living room tuner, the Pioneer F-26.

I'm adding it because it seems to sound better.  I have reasonable ideas why it might sound better also.  The Pioneer actually has a fairly high impedance output.  Even though it has separate fixed and variable outputs, they all originate from the same opamp, the "fixed" outputs (which most but not all audiophiles assume is better on most source devices) go through a voltage divider with 43k on each side, resulting in a source impedance of 22k (!!! this has always appeared to me as a deliberate way of creating high frequency rolloff), the the variable outputs go through a pot in series with a 330 ohm resistor on the ground side, resulting in a minimum of 330 ohms minimum perhaps, with fixed all the way up.

I had first assumed then primitively tested the fixed was better, and I might have tested it too.  So since my usual connection was to Sonos, I ran the fixed outputs to Sonos.  So when I connected first to the PMD 580 (whose conversion to digital sounded horrible to my ears) and then to the Lavry AD10 (which sounds wonderful) I was using the Lavry.  I have the "fixed" output all the way up, so it should be an output impedance of 330 ohms and a little more.  This is fairly high as solid state equipment goes, and could cause high frequency rolloff with long runs of some kinds of cable, and destination equipment.  But it seems it could be far worse with the fixed outputs (which raises some more interesting questions, perhaps a less is more freak would find some adjustment of the output control yielded "just the right" sound by affecting the high frequency cutoff.  but achieving a similar effect with an external filtering and buffering active device could actually have many advantages, including higher ultimate slopes of high frequency cutoff, combined with steep low frequency cutoff, and loading the originating op amp circuit including the output capacitor the least possible).  Interestingly, the cutoff from the fixed outputs at 22000 ohm output impedance into a typical 100 pF of cable capacitance plus other stuff would be 72kHz.  This is not far from the high frequency cutoff of the X-10D.  So using the fixed outputs without the X-10D would have about the same high frequency cutoff as using the variable outputs with the X-10D...

I'm thinking that the X-10D serves a function similar to the "Noise Filter / Buffer" MA Cotter NFB-2, which was designed to improve on the output stages of nearly everything.  In addition to providing a very high input impedance and a very low output impedance, the NFB-2 was claimed to remove out-of-band noise without adversely affecting the audible spectrum.  This was said to reduce the demands upon later equipment, thereby enabling it to perform better.

"Sure, every equipment maker could have an output stage as good as the NFB-2 on their equipment.  Fat Chance." said Peter Aczel of the Audio Critic before he became a full audio objectivist possibly eschewing such differences.

What was in the MA Cotter NFB-2 ?  It was never revealed to the public, and I've never been able to find out online.  They were marketed to and sold by the audio salon and modification studio I worked at known as Audio Directions / Audio Dimensions.  I believe I saw Mitch Cotter himself demonstrating the products to the staff, including co-owner Ike Eisenson, who then took Mitch to the back room for a long discussion.   A senior technician later remarked, "It probably has Nuvistors in it."  Ike was a big fan of Nuvistors and wrote glowingly of them in the Audio Dimensions magazine and Tu-be modification manual.  FWIW, Tony Michaelson was in our orbit as well.

Interestingly enough, the milspec pencil tubes in the X-10D Version 3 are very similar in characteristics to Nuvistors, ,the main difference that the pencil tubes are made of glass rather than the Nuvistor metal.  The pencil tubes are similarly compact, which makes them similarly (if not quite as) robust, immune to vibration and long lived.  In both cases projected life can be around 100,000 hours, which is at least one solid decade of running continuously, and possibly more in intermittant use.  Playing up this similarity, Musical Fidelity called them "Mu Vistors," after years of making equipment with actual (but very limited supply) Nuvistors.

"tubes" as we call them intrinsically have very high input impedances, and generally have sufficient inter-electrode capacitances (combined with internal high impedances) to have inherent stability and limited response.  It's also easy to limit the low frequency response with small high quality film capacitors before, after, and mid-stage, because of the high impedances.  Cooked up just right, or at least like the X-10D Version 3 has them, tube resistance coupled amplifiers naturally work as buffers and as out-of-band filters!  You don't even need to add anything else.  The very nature of its operation filters out-of-band signals.

Now people might ask, why not use tubes for everything?  The answer may be, that tubes are not necessarily the best for everything.  It is hard to make good power amplifiers with tubes unless you are going to use an output transformer...and that's a huge limitation.  And so on.  They just happen to be the best for this one role of "noise filter / buffer" because it's a natural one for them.

So it makes perfect sense to have tubes in one and only one spot in a high performance system.  And that is as the high performance output stages of especially noisy equipment, such as tuners.  As a noise filter / buffer for the preceding stages.

Interestingly enough, Mitch Cotter also worked on the famous Marantz 10B tuner, which has wonderful sound by many reports.   People generally look to the impressive 6 tube "Butterworth" (not really) IF stage, which not even Marantz has even attempted to duplicate.  (Marantz did include similar in concept IF stages in the 20/20b/120/150 tuners but using transistors instead of tubes.)

But one thing also, the Marantz 10B also has a tube output stage, and that could explain some of the "good sound."

Now what I'm suggesting here isn't at all a new idea.  Musical Fidelity cooked up the first X-10D in the 1990's, and for much longer than that people have been sticking tube outputs on nearly everything you can think of that may use no other tubes.  Sometimes they add tubes just for a light show.  There was one guy who did modifications where he stuck tube outputs on nearly everything.  He went by the name Lampizator.  And now there are endless Chifi tube buffer devices you can buy on eBay.

It's only radical for someone like me, who never used tubes for everything, and finally abandoned them altogether around 2002, and whose closest friends haven't used tubes since the 70's or earlier.

Tubes are widely perceived by non-tubeophiles like some of my friends as "coloration devices."  But a well designed tube noise filter / buffer can be "perfect" by audio objectivist standards, adding no perceivable audible coloration according to audio objectivist interpretations, and still improve the sound by reducing out-of-band noise, or reducing impedance mismatches.  Generally speaking, tube equipment can be designed to meet every specification as transistorized equipment, even power amplifiers (but at much larger cost).  Ancient tube equipment did not often do this (but see, for example, Western Electric amplifiers, which were very good).

It might, in some cases, add an audibly euphonic amount of noise.  That might actually help in some cases (similar to dithering noise used in digital).  But it would have to be very small to be considered "not a coloration."

I think a tube noise filter / buffer for a high quality system needs to be designed to a very high level of performance.  It should have at least 20-20kHz bandwidth, and 20-20khz distortion below 0.01%.  The X-10D V3 is claimed to do better than that, 20-60kHz and 0.004% distortion.  That looks good.

The bandwidth, signal to noise level, and dynamic range requirement may depend on what equipment it is being applied to, the type of music being played, or other factors, but weighted noise should probably be below -100dB of peak level.

Most people who are interested in such things eschew all specifications in the first place, and maybe even think devices with around 1% distortion or so to be "ideal."

In my limited experience with such things, amplifiers with 1% distortion or higher sound awful.  I don't think even high order harmonics are audible below 0.01%.  Few power amplifiers have distortion that low, but their harmonic profiles may be relatively innocuous, mostly weighted toward the lowest orders.

Now a general purpose device to do this might have settings for bandwidth and noise level.  The high frequency response could be varied with an air capacitor adding to appropriate elements, varying the high frequency cutoff from 200kHz to 20kHz.  I don't like the sound of low pass filters that cut below 20kHz.

Because of their inherent stability, feedback need not be eschewed with tubes.  Or with well designed transistor equipment either.  But many tweak designers still live in fantasy land, and zero feedback, which typically yields too much distortion to be inaudible distortion, is deployed.

For many years I personally believed that resistance coupled tube amplifiers (like the PAS-3X) should have low impedance cathode follower outputs.  Higher end preamps like Marantz Model 7 tended to do that (though IIRC they included feedback around the cathode follower too).  But many audiophiles long believed cathode followers had a bad sound quality.  While the cathode follower is a simple and stable circuit, it cannot be made to have as flat frequency response as a dual triode stage with loop feedback.  Or as uniform output impedance with frequency.  The more "complex" dual triode with feedback has long been loved.  The feedback works to overcome the tube imposed bandwidth and impedance limitations with the restraint (the resistors and capacitors of the feedback network) being almost perfect as compared to the cathode resistor which is one step removed from the actual output.  The PAS-3X line amplifier was legendarily good in itself, and the subject of endless additional modifications.  Very few added a cathode follower--even though it is a rather high impedance circuit measured in kohms as compared to the 33 ohms of the X-10D Version 3, which achieves that due to the type of tubes and high degree of feedback.  The high frequency cutoff is limited to 60kHz instead of the 100 kHz of a 12AX7 or 12AU7 circuit, but as a noise filter buffer for an FM tuner that may be a good thing...maybe even lower.  I'm not sure exactly where the cutoff should be to be transparent, but 60kHz seems like a good start and work down from there.

I'm now planning to implement an even more ambitious maximalist concept.  Something I've had in mind for a long time, but I've never had enough time to do it correctly.

Automatic background music.  Whenever the system is silent for more than 50 seconds or so, the FM tuner will drop in as replacement background music.

One of the key issues here is that regardless of what level was chosen for the previous input, which may even have included above 0dB digital gain (for low gain sources, such as multichannel movie discs where "summing" is used to generate 2 channels for the living room system), the tuner should switch in at a background music level, or at least no louder than I would like to listen to the tuner, which is somewhere between -10dB and -23dB because of high volume compression on FM, even on the classical station.  You don't want to listen to FM as loud as other sources because the FM is continuously pushed up to the broadcast limit.

But now I have a perfect way to do this.  I have already wired this up: the FM tuner is wired to a dedicated Analog to Digital converter.  It would be nice if I had another Lavry AD10, but I bought something cheaper and nearly as good a few years ago, a Black Lion Sparrow, the original which has no level controls.  Fortunately, the sensitivity (claimed 1.6V) is perfect for my Pioneer F-26 at the more transparent sounding variable output turned all the way up.  I was worried I'd have to turn the F-26 down, but in fact it's already only hitting about -8dB, could even be cranked up a bit, but the Black Lion is a very quiet (-120dB claimed "A" weighted) converter.  And that low level helps me achieve the desired background level of -23dB in combination with the maximum -15dB attenuation of a Behringer Ultra Curve 2496 DEQ  (I wish Behringer had allowed far more attenuation, and in 0.1dB increments for each channel).

So the new setup is like this:  The variable output of the F-26, cranked all the way up, feeds a Musical Fidelity X-10D V3, my latest minty one (looked new in box), which feeds the Black Lion Sparrow (Using RCA to mono 1/4 plugs as an unbalanced to balanced adapter), from which the AES output goes to the Behringer DEQ, which provides some EQ tweaking capabilities and up to -15dB reduction, from which the AES output goes to a Hosa AES to unbalanced SPDIF adapter, and thence to a 4-way video switch (on which the video jacks have sufficient bandwidth for SPDIF) which feeds digital input on my Tact digital preamp.

The plan was to get this new digital path sounding about as good as the one where the X-10D V3 feeds my Emotiva preamp and thence my Lavry AD10 converter.

I think the setup above does this very closely, with only a bit more bass overhang than the Lavry path, and that could possible be adjusted (I'm already doing so to a limited degree) with EQ adjustements in the DEQ.

Also important, the attenuation reaches the required level that I can feed the digital signal directly to my crossovers without attenuation.

So now, I can get a 3-way electronically controlled AES switch.  One "way" connects to the output of the Tact for everything except FM.  The second "way" connects to the new FM-tuner-to-digital-output path described above, except directly from the AES output of the DEQ (no conversion to coax SPDIF is required).  The third way is not connected to anything, making it a "mute" selection.

To that, I add a "silence detector" at the analog outputs of the Tact--which I never routinely use--which causes a contact closure or open after 50 seconds of silence which either controls the AES switch directly, or feeds an Insteon low voltage interface to send an Insteon home control signal, to which another Insteon low voltage output interface communicates with the AES switch.

Running the whole thing through Insteon lets me add buttons elsewhere which enable the "mute" selection on the AES switch, in case even the FM is bugging me too much, or switch back to FM or the main program.

But I find the new digital conversion on FM, and also listening at lower levels like -20dB from peak, cause me to feel like muting it less.

There are many times when you want information loss.

Adventures with FM Radio

FM radio was my first introduction to High Fidelity Audio, and I still love it.  Local radio stations play local live performances and other music that can't be obtained elsewhere.  Professional DJ's curate streams of music from endless sources that rarely grow tiring (on the mostly noncommercial stations I listen to).  You may not love FM still or yet, but still I think I've made some interesting discoveries investigating FM radio that may apply in other circumstances.

My chief way of listening to FM radio now is the Kitchen Tuner, because it is connected to the highest antenna I have now, the MD whip antenna at the peak of the roof, with grounding installed by licensed electrician.

But more about that later.

This really starts with playing FM in the living room.  20 years ago I found an amazingly good indoor antenna location on the front wall of the house, near the ceiling and the door, for KPAC, the local noncommercial classical music station I listen to most.  Curiously, this one location was far better than anything else in the house.  As much as I preferred to listen to FM in the bedroom, having a tuner in the bedroom was never very useful, that seems like the worst location for an FM antenna, which is all the more curious because it's actually closer to where my favorite stations are being broadcast from.

Since all my audio sources must feed a digital processing network, analog sources like FM must be converted to digital.  For a long time, I've had many ways to do this:

1.  The Tact 2.0 RCS digital preamplifier I have has the analog input board, which converts analog sources up to 1.6 volts full scale to digital, with approximately 16 bits of resolution according to John Atkinson.  I have generally avoided these analog inputs except for testing purposes.

2.  I've played and streamed music to the living room system through Sonos since about 2006.  I currently have a Sonos Connect node as one of my main digital sources to the Tact.  Sonos ZP80, Connect, and similar nodes have line-inputs, which you can set to "uncompressed" and then play them on any other Sonos playback jack or device.  This was one of the #1 reasons I got Sonos in the first place.  So I could listen to the living room tuner, which had by far the best antenna inside the house (because of location, and for no reason I've ever been able to figure) in the bedroom, and later other rooms, including the living room itself.  The analog input on the Sonos seems to be better than the one on the Tact, and can take well over 2V inputs.

3.  Through the Lavry AD10 I purchased in around 2008 or so, when I started getting serious about the Living Room System.  The Lavry is a $1488 professional digital Analog to Digital converter, with 24 bits nominal resolution, and dynamic range of 117dB dynamic range (120dB weighted) and distortion below 0.0009%) designed by the analog to digital converter designer famous for having previously been lead designer on the legendary Pacific Microsonics Models One and Two.  It wasn't his ultimate effort at the time (or since), that has always been the Gold series, and this is only the Black series.  The Gold series had about 10dB more dynamic range, or now even better.  The Gold at the time I bought my black was legendary, based on real self-calibrating resistor networks, though far more complex than mine and possibly requiring more service.  My AD10 has been absolutely reliable and I've always expected it to stay that way.

However, this approach to listening to FM had a downside back for the 10 years or so I was using my Krell FPB 300 mostly.  It was first with FM radio sources that I experienced a mysterious "shutdown."  This may have been just coincidence but I figured FM radio had curious DC shifts that drove the Krell into unsustainable high current class A modes when not much was actually being delivered to my Acoustat speakers.  But I don't really have any explanation for it, I just basically rarely used the Lavry AD10 to listen to FM radio.  I considered it dangerous.  So it hadn't really been in mind much recently, even though I now use amplifiers that are perfectly reliable and never shut down, especially the Hafler 9300, designed by Dr James Strickland--who also designed the Acoustats, and also the Aragon 8008BB.  As it turned out, the problem with the Krell shutting down was a problem with the Krell itself, that I tried to have repaired, but I now believe stems from a combination of over-aggressive biasing and the lack of availability of the original transistors.  If I had been lucky, I would have gotten Transistor Service in the second channel at the time they did the first.  Now it's impossible to get that service, so it has one good channel and one problematic channel.  My long term dream is to reprogram the computer for lower and safer bias levels.  But even though I'd eliminated the shutdown problem by moving on to other amplifiers, I was still avoiding listening to FM radio through the Lavry.

Now, this works very differently generally than people surmise.  People think that because you may start with what they perceive as a "low resolution" source like FM radio, it therefore doesn't matter so much how you convert to digital, because digital is so much better.

But this is to fail to appreciate that the noise and junk that is included in the output of an FM tuner (which might be similar to a phono system in these regards) is best reproduced--or attenuated--perfectly.  If the crud isn't reproduced perfectly, the newly mangled crud is even more objectionable.  The upshot of this is that it may be just as important--if not more--to encode crappy sources in high resolution as better sources.  This is a general principle that goes beyond digital encoding to all forms of signal transmission.  Crappy sources must be reproduced better to sound decent.  It's also possible to attenuate the crud, but that must be done nicely rather than crudely.  This is the opportunity where a well engineered noise filter and buffer might be useful.  Classic FM tuners, even my legendary Pioneer F-26, was a rather mediocre output stage (which I've sometimes wondered already does some of the filtering that makes it sound so good compared with most other tuners).

4.  Recently, to time-shift programs on FM radio, I purchased a Marantz PMD 580 digital recorder.  This was the final step in a long series of time-shifting arrangements.  Until 2020, I was recording FM radio programs on my cassette recorder, an auto-reversing Nakamichi 550 (once again, the more reliable second down model from the Dragon--notorious for needing costly repairs).  So the way this worked is, programs were played on my living room tuner, whose output went into the Sonos line input, through the Sonos network and out through the ZP80 in the bedroom #2 position, which I've dubbed "Turntable and Tape", and direct into the Nakamichi.  I used Chrome Type 2 tape and Dolby B.  Some of the recordings I made by this ridiculously complex setup sounded remarkably wonderful, in fact I felt that almost all of them sounded quite good.  Somehow the Cassette Tape recording filters the FM in such a way that it sounds even better, or so I think.

But it still irked me that I could not make recordings directly to digital, which is virtually lossless.  I originally planned to make such recordings on the Masterlink digital recorder in the bedroom.  But it refused to record from the Sonos digital output, claiming "Copyright Violation."  Sonos apparently sets that flag for all analog input sources.

5.  At least I figured I could record the analog itself directly on a new Marantz PMD 580 I bought this year for the living room.  I mainly intended this as a replacement for the Nakamichi 550 in FM time shifing.  I figured it HAD to be better.  I was bypassing a whole serious of Sonos conversions, first to-digital and then from-digital.  And then recording on cassette tape, an obviously fidelity limited system (but pretty damned good sounding when done well, I must add, and especially for FM radio sources).

I could the play back directly to the living room system as well, and in nominal 24 bit 48 kHz quality.  Not the 96kHz I prefer, but I think 48kHz is nearly as good, and the 24 bits is the more important part anyway.

I could also play live radio directly through the Marantz, with the recorder in pause, letting it do the digital conversion.  This ought to be better than the Sonos conversion by being 24 bit and 48 kHz.  As well as playing back time shifted music.

It sounded very clear with no added noise and reasonably good.  But it did have a sort of exaggerated sibilence that digital is mostly unjustly now infamous for.  It sounded electronic, hifi, etc.  I was lusting for my old Cassette Tape playback, even coming back from the Cassette Player once again through Sonos to the Living Room.

So then, it occurred to me to try the Lavry again.  I've put Emotive XSP-1 Gen 2 unit number two, the nearly new one, into the Living Room, recently.  I knew I had Input 2 available, and it was fairly easy to hook it up to the tuner Variable Output (I use the Fixed Output for Sonos) with gain turned all the way up.

The sound was so much better, it was an immediate revelation.  I will never go back to the unbalanced analog input on the Marantz again.  Using the Lavry through the Emotiva is just worlds better.  No "digital" sound at all, just more like it's not there.  No argument, really, we can see it should be from the specs involved. The Marantz has 20dB or so less dynamic range (S/N is rated at 91dB) and higher distortion (rated at 0.01%).  I think those are more important than the additional fact the Lavry is digitizing to 96kHz rather than 48kHz, but that's one more detail.

But, but, I was inspired to go further.  There was still a bit more boom, pop, and "transistory" sound to be that short of perfection.  I was inspired to try something very different.

I was inspired to try one of my rarely used Musical Fidelity X10-D V3 tube buffers.  I was inspired by how good the cassette recording sounded, as compared with the Marantz.  Perhaps the output of my Pioneer F-26 FM tuner needs a bit of conditioning before digital conversion.  It had really been designed to feed into analog preamps, not digital converters.

My current feeling is that this is exactly correct.  The X10-D in the above system, prior to the Emotiva, makes it just that much more pleasant.  I feel good enough to keep listening longer, and even to other stations I'd hardly bother listening to.

It seems in particular the X10-D V3 removes some low frequency out-of-phase garbage and very low frequency instability and popping from the tuner.  Excess pop is greatly reduced, so I can crank the level up more.  Also the highs sound more relaxed, with a reduction of apparent sibilence.

These are the kinds of things you might or might not expect from the V3 specs, 10 Hz-100kHz.  Note that John Atkinson measured actually a slightly rising response to 10Hz and extension considerably beyond 100kHz for the first version.  However, we know that somewhere before DC, the X10-D by necessity (as the RC coupled tube amplifier that it is) adds 3 6dB/octave poles on the downside, and something like that on the upside.  That filters out very low frequencies close to DC, the flapping around that FM tuners tended to have in the old days when they were good.  The X10-D is by necessity a filtering device.  However, the filtering is way outside the 20-20kHz nominal range.  Is that important?  Yes, I think it is.  I wish more digital processors had controls that allowed you to set cutoffs at very low or high frequencies, as that is often more useful than prosaic filters at audible frequencies.  The crud is often best filtered out with out-of-band filters, and that helps reduce later in-band effects.  In band filters typically make matters worse more than better.  Somehow I can do the in-band filtering in my head better, just not the out-of-band filtering.   It seems your head can remove the in-band noise better than a filter, but less so for the out-of-band noise.  Or perhaps that's just my imagination.

Atkinson also measured in the original X10-D a loss of stereo separation at very low frequencies.  He blamed the power supply.  Many changes happened between that model and the one I have.  But if some of that loss of stereo separation also exists in my unit, it might also help the FM low frequency rumbling and popping.  I had an insight "maybe this thing was actually designed for FM radio, achieving effects similar to a output stage on a premium tube tuner that sounded good, but then they tried to sell it for the more popular digital systems."  I had never found it to be useful on digital system (though, perhaps, I never tried much either).

Reducing the load on FM tuner outputs to a negligible 300K is also desirable.  This prevents the tuner from wasting any output current, avoiding any kind of rippling in the power supply.  As well as reducing distortion from the output circuitry, and improving bandwidth, which are automatic with lower impedance.  I've always thought the reduction of sense-of-strain (distortion and delayed affects through power supply) to be more important than the increase of bandwidth when input impedances are lowered on less-than-high-current outputs.  But I now figure this to be imporatant factor on most analog connections.  Input impedances 100K ohm and higher are desirable.  This may not be a factor with $10K 2000's vintage Accuphase and the like super tuners, which may have high current output stages of comparable quality to the reset of the tuner. 

I'm not sure how well I'd do proving this with blind experiments.  I have wired up a switch to switch back and forth, and my tendency is to stick with the X10-D V3 circuit.  I have to simultaneously remember to correct for the 0.9dB of added gain that the X10-D provides, and assuming my V3 has a similar amount of gain, by flipping the gain on my Emotiva by 1.0dB (I wished, like Levinson, the Emotiva had 0.1dB level adjustments for audio reviewing purposes, and could memorize them for particular inputs...nobody but Levinson does such things, instead Emotive spent big money on home theater tricks I never use...I'd like polarity mono and stereo reverse controls too, but unwilling to spend the big buck on Levinson, which doesn't even spec as good as Emotiva until the top stratospherically priced Levinson models either...but I continue to lust for a ML 32.)

Endless tube buffers available on ebay from Chifi companies might serve a similar noise-filter/buffer as the X10-D V3.  But I don't think many if any of them are engineered to the same high fidelity standards, with vanishingly low distortion, and essentially flat response in the "audible" range.  People are more interested in gimmicks like being feedback-free.  Feedback-free tube circuits inject familiar tubey colorations.  I'm not interested in those anymore.  The X-10 V3 is basically neutral and is a very high fidelity device.  The noise of the X-10 V3 is even lower than that of the Marantz PMD 580, plus it is nice analog noise, not correlated and awful digital noise as is generated by the Marantz but suppressed to total inaudibility by good converters like the Lavry AD10 (which are invisible to me).  I think there were issues in 90's era digital encoders, where they thought they were being smarter with one-bit and the like systems which measured better.  But they weren't being smarter, it was a case of too-little-knowledge, and later sigma delta systems up the the present are far better.

So, along with all this transformation of FM in the living room, I'm headed towards a similar transformation in the Kitchen.  I moved out the Sansui D99X, a rarely recognized sleeper that actually represents Sansui's last and greatest digital tuner.  (It was highly recommended in an Audio Magazine review.)  It uses Sansui's proprietary Walsh decoder--which they invented.  Just before the company went bankrupt.  The tuner looks cheap, identical to a $49 model (but heavier).  But it's special, as it turns out.  Very very quiet, noise just completely disappears nearly always.  Sansui clearly uses a few tricks to achieve this sound, as they also did in the preceding X-1, as well as superlative FM engineering.

I had just moved that into the kitchen to finally test it.  I was planning to donate it to Goodwill.  It had been in my collection for 15 years or so, but never used.  It looked so boring, compared with my other super tuners.  And nobody ever talks about it, I just happen to know the history from reading old magazines, including the article written by Sansui Engineers about their new Walsh Decoder in the Journal of the Audio Engineering Society.

But once it was clear how good the Sansui was, I wasn't willing to move it out again.  I did recall that the Kenwood L-1000t it was sitting atop was an even more ultimate supertuner, being the last super tuner made by Kenwood, whose design was praised by David Rich in the Audio Critic.  Problem is, my L-1000t has a horrible drift problem, so bad I wonder if James Bongiorno wasn't getting confused when he said the Sony 730ES has a horrible drift problem (I've never noticed any in mine).  After about 6 hours on time, you must re-tune every station by 0.025 Mhz (which is possible on this tuner) to keep stations from sounding distorted.  But when the tuning is fixed like this, the Kenwood sound so fabulous you could confuse it with CD, perfectly transparent.  By comparison, the Sansui, as well as most other tuners, including super tuners sound dark.  Kenwood tuners generally sounded too bright, but the L-1000t gets it exactly right, for once, in their final super tuner.

I hadn't listened to the Yamaha TX-1000U in ages, but I remembered liking it.  It's essentially Yamaha's last super tuner, though they made a virtually identical TX-2000U in a fancier box.  It was never appreciated as much as Yamaha's legendary analog tuner, the CT-7000, but other than using the digital front end (which simply cannot be as good as an analog front end) it's more advanced in other ways, using a proprietary analog multiplier for FM stereo decoding, the ultimate best way of doing it, but only copied by a few other companies.

The Yamaha sounds damned good.  Both more lively and more earthy than the Sansui.  We've moved out of the shade and into the light, but still noise does not intrude.

I hope to figure out how to fix the drift in the Kenwood.  I cleared my bench in the first time in a year (it was plied up with junk after my Acoustat repairs and a few others last year) and deposited the massive L-1000t on top of it.

As long as I'm using the Yamaha, or the Sansui, or many other tuners in the kitchen I also need an X10-D V3 for the kitchen, where I'd like to send one copy of the tuner audio signal to Sonos to enjoy elsewhere in the house much as I do for the living room tuner.  This works out terribly using Y adapters, I've determined, because most tuners sound much worse into a lower impedance.  This is another perfect spot for an X10-D.  Whichever output sounds better for listening in the kitchen will go to the Yamaha or future home theater receivers, and the other output sent to the sonos, so their loads don't add up.  The Yamaha has 47k impedance and Sonos about 38k, but combined that's getting close to 20k, which is undesireable for most analog tuners.

I had two X-10D V3's but only one power supply.  The second was an eBay mistake from years ago.  I thought I could just buy a 24VAC supply, which I did, but then found you needed a center tapped 24VAC.  I just bought an upgraded ChiFi 24V supply designed for Musical Fidelity products.  It will be interesting to see if this actually makes the X10-D sound better, or perhaps some of the "goodness" on FM is due to the relatively limited 500 MA power supply imposing limits of its own and thereby reducing the "popping."

The specs for the X-10D V3 show a lower high frequency specification (60kHz) than the original X-10D (100khz).  For "noise filter/buffer" purposes, the lower cutoff may actually be better.

It was Mitch Cotter who designed and sold a "Noise Filter Buffer" audio device (the M A Cotter NFB-2).  I was a technician at a high end store when it was introduced--it was considered a very big thing.  It was primarily intended for use in a phono playback system.  M A Cotter also produced turntable bases, preamps, and moving coil cartridge transformers.  We used all of those in our store.  However, the NFB-2 was not marketed especially as a phono correction device, but as a device needed by all good audio systems to remove noxious out-of-band signals.  Mitch Cotter was a technical consultant for Marantz for the development of the Model 10 tuner.  

The M A Cotter devices were very costly when they were originally sold in the 1970's, and still today.  Sadly there seems to be absolutely no technical information about the devices available, such as cutoff frequencies, slopes, or other details.  It looks like a blue brick, and I wouldn't be surprised if the circuit on the inside was potted.  To operate one of these, you not only had to buy the device itself, but also a large power supply unit capable of powering 5 devices. Today the power supplies are the rarest and most expensive items of all.   Cotter must have had some good ideas, but they are all pretty much lost to most people now, except the very rare users of his devices.  But since he had previously worked on the 10B tuner, noted for good sound, it's possible he started thinking about noise filtering and buffering with FM tuners.

The original X-10D had two tube sockets and you could try different tubes.  The X-10D V3 has a "pencil" 6122 tube soldered in.  These tubes were designed for missile navigation, are very rugged and rated to last 100,000 hours.  That's almost worth the bother of soldering them in.   The V3 had very slightly lower distortion along with the lower 60kHz cutoff.  It's easy to imagine the smaller tube has slightly higher inter-element capacitances and therefore the lower cutoff.  Musical Fidelity may have bought the last set made by Phillips, but there are other versions.

I was inspired to buy a third minty looking X-10D V3 so I can have a spare unit for experimentation, repair, and possible modification.  I currently imagine have one in living room and one in kitchen for the tuners in those locations.  They work for either sound improvement, "splitting", or both.  Even if a tuner provides two outputs (as some do, "Fixed" and "Variable" usually) it is generally the case that one of those outputs is better.  With an X-10D V3 I can split the good output two ways to go to the local system and also the nearest Sonos Connect input to be available in other rooms, and all without worrying about excess loading on the tuner  And the use of the "buffered" and "unbuffered" outputs on the V3 will also be decided in which I consider most important, but likely with the better sounding output going to the local system since the other is just going to Sonos anyway.

My Linn Sondek LP12 Valhalla has accurate speed

Back in the early 1980's a friend of mine visited a high end audio salon with a test record and a guitar tuner--a handheld device which has a needle meter showing how flat or sharp a particular musical note is.  He tested a bunch of turntables and told me that they all ran at the wrong speed.  Especially the Linn Sondek, he added.  The only belt drive with accurate speed, he said, was the Music Hall MMF-5.  Meanwhile, he claimed, all the quartz locked direct drive turntables were correct, and especially his Sony PS-X800.

I didn't believe him then, and I think now he was at best misinterpreting or overgeneralizing now, but it has remained a nagging fear.  My friends probably bogus claim has given me a largely unwarranted fear of turntables playing at the wrong speed ever since.  Another variant of audiophilia nervosa.  As teenager I leaned in the opposite direction.  My very first turntable was a Dual 1209 which had a 6% adjustable speed control that changed the position of an idler wheel.  It should probably have been adjusted for accurate speed using a strobe light.  It came with a tiny strobe disc.  I never did that adjustment, I considered it unimportant, and the required neon light seemed unobtainable in those days before online.  Using my Tensor lamp was useless.  I lusted for a Dual 1219 which had the strobe indicator built in and much more readible.  For years I routinely listened with a 1% or thereabouts speed error which I never thought much about, except being pissed that I hadn't gotten the 1219.  I figured even a 1% error wouldn't make that much audible difference in a playback situation, but I hated not being able to brag about it.  I still haven't actually explored the audibility of these kinds of differences.  Such an investigation, done correctly with blind listening tests, might take forever because the change is so small (unless you have perfect pitch, or maybe even if you do).  A semitone is a 5.9% increase or decrease, so a 1% error is about 1/6 semitone.

My friend argued that the new turntables with quartz locked motors were the only kind that could really maintain the correct speed.  I argued that a belt drive turntable with synchronous motor stays locked either to line frequency or a precision quartz oscillator, and it basically cannot have overall speed error (just wow and flutter) unless the pully is manufactured to the wrong size, which would be a serious manufacturing defect.  Any decent belt drive turntable with a synchronous motor ought to have the correct speed.  (I found out much later that some early Rega models, from the early 80's and before, did have such a defect--aka "feature"--and ran slightly fast, a problem which was corrected after many complaints.)   Meanwhile, a direct drive turntable with a DC motor has to use a servo just to maintain any constant speed, since otherwise the speed is not fixed but a function of current and loading.  It has to have a low voltage reference oscillator and the only way to make that reasonably accurate is with Quartz.  Meanwhile the frequency of the AC current used in a synchronous motor is already sufficiently accurate.  My friend said well then maybe the Linn wasn't really made that well.  He was not the first to say that based on what I know now to be misunderstanding.

I went with my friend (who also says he has perfect pitch) to that high end store sometime later and he showed me the guitar tuner needle going up and down on a particular turntable.  I couldn't explain the up and down at the time, but I pointed out that the going up and down seemed to center around the correct speed.   Yes, he said, but see how it goes a little higher than it goes lower.  It was hard to tell actually, sometimes it peaked higher than it dipped, and other times it dipped lower than it peaked, and the ultimate speed accuracy wouldn't depend only on how much higher or lower it goes on one particular round, but how much time it spends higher or lower, which I couldn't estimate, I said.  He argued that it couldn't go any higher than it went lower, and that I was just rationalizing.  (I have subsequently figured out why it might go higher than it goes lower on any particular round, and vary a lot in between, read on.)

A few years ago I got the Feickert Adjust test record and the PlatterSpeed app for an android phone.  (Nothing like that app seems to be available for iPhone.)  Now I could do what my friend used to do, but with the kind of accuracy required to do it correctly.

I applied this test to my Mitsubishi LT-3 turntable (which was my only working turntable at the time).  I saw right away (as well as heard) the up and down in the 3150 Hz playback tone.  The up and down aligns perfectly with record rotation and represents the effect of an off-center hole in the record.  It turns out that all LP records have off-center holes.  It's apparently impossible to fix during record manufacturing.  There have been a number of audiophile products to correct this well known problem, including ways to punch new center holes by hand, and turntables (a couple of Nakamichi models) which automatically re-center records for playback using lasers and computers.  Bob Green of The Absolute Sound has written about how important this problem is, and complained that hardly anyone pays attention to it.  And yet, it might be one of the most audible defects of LP playback generally.  The off-center wow in virtually every LP far exceeds the wow caused by most turntable platters themselves.  Various WOW specification weightings eliminate the effect of this very low frequency wow, and it is generally considered more audibly innocuous than higher frequency wow caused by the drive system and bearing.  I've sometimes wondered if it adds a euphonic coloration that turntable lovers actually prefer.  You can see the off-center wow prominently in turntable tests published by Stereophile.  Stereophile reviewers look past the off-center wow to see what the actual platter wow looks like, because that is what actually varies between turntables.  Off-center wow is a constant determined by the records you are playing.

The PlatterSpeed app does what I couldn't do by eyeball.  It averages the speed over many seconds and gives you a digital readout of the exact average speed.  If I remember correctly, the Mitsubishi did have very close to correct speed.

As I was getting my Linn Sondek LP12 repaired a few years ago, the Linn guru Mark (a former dealer) did the speed adjustment AFTER the turntable was set up in its intended location.  He said that was the time to do it.  I happened to have a $30 optical test record, and we used that to perform the speed testing.  The speed on an LP12 (except for the newest models) is adjusted by adjusting the 3 motor screws.  The motor pully is either tilted more towards the turntable, or more away from the turntable.  It can pretty much be set as accurately as you have patience for.  He got it accurate enough that you couldn't see any of the markings on the test disc advance or decline over 30 seconds.  That was as good as we had patience for.

At first I thought maybe the distance of the pully from the platter was being changed.  But a friend convinced me that would not make any difference.  The only thing that could vary the speed would be varying the size of the pully the belt goes around, and the size of the subplatter the belt goes around.  Nothing else should matter.  The motor speed is locked to a precision crystal oscillator on the Valhalla.  On other belt turntables, the speed is taken is taken from the AC line, which has incredibly accurate frequency which can't vary more than a second over many months because power equipment and clocks depend on that, but considerable noise.

So then it bothered me because the method of speed adjustment on the LP12 seemed like a hack.

But now I've determined that the motor screw adjustment on LP12 is not at all a hack.  It deals with a fundamental problem with a suspended-subchassis type turntable like the LP12.  The problem is that the subplatter might not line up perfectly with the motor, since the subplatter is supported by a suspended subchassis which can move up or down and even slightly to the sides.  As the belt ages, or the springs age, this alignment may shift ever so slightly.  When the belt wraps around the motor pully at a greater or lesser angle than 90 degrees, the effective radius of the pully changes slightly.  The motor screw adjustment fixes this by restoring the pully axis to 90 degrees from the angle of the belt.  It actually fixes the problem as close as possible to the source of the problem.  Or you could re-tune the suspension, but that is notoriously difficult on a Linn Sondek LP12 and best left to a Linn expert.  Tuning the speed with the motor screws is much easier, and it can easily be dialed in as accurately as you choose.

Mark told me I should re-check the speed every year because it could be affected by belt aging.  I didn't understand that at the time, but later figured that the belt itself has some influence on the suspension.  As the belt loosens, the suspension can tilt slighly away from the pully, affecting the speed.  The pully can then be re-adjusted for the new tilt.

Simpler belt-drive turntables which lack the suspended subchassis don't have this problem.  They can stay in alignment forever.  But then what you give up for that is potential motor and bearing noise and acoustic feedback.  The Linn LP12 is one of the quietest turntables made primarily because of the suspended subchassis, which was an idea invented much earlier and originally popularized by the AR Turntable.  Using a suspended subchassis, motor noise can be reduced to nil, even with inexpensive motors.  And acoustical feedback can be very much reduced also.

But anyway, it is possible that the LP12 my friend once measured hadn't been adjusted in awhile.  The store he did these tests at was known for being a bit sloppy, stuff often didn't work right.  Or it's still possible that it was just his inaccurate "eyeball" method of estimating the average speed.

So now, a couple years later, I thought it would be good to re-test my LP 12.  This would be an indication of how much speed drift there might be in the first two years after changing the Linn belt, when it probably changes the most.

Understanding as I do the way the Linn suspension can affect the speed, I realized I should test records of different weights to check that out also.  The Feickert test record itself is 7 inches and very light, but I figured I could stack at least one record underneath it.  (That's about all I can do.)  All the tests included the Feickert adjust LP itself, combined with another record and/or clamp.

The record has a 3150 Hz tone.

Test Condition                        Avg Speed        10 sec Readings

Heavy LP and clamp               3150.7              3150 3150 3150

Medium LP and clamp            3151.8              3150 3151 3150 3151

Feickert plus clamp                 3152.2               3152 3151 3152 3151

Feickert w/o clamp                  3152.7               3152 3152 3151 3152

This shows the small increase in speed as the weight of the LP increases, just as I suspected.  I'm not sure what the time interval for the "Average" speeds are and whether they are actually better or not than the "10 sec" readings, but for the purposes of this analysis I am using them.  If I were writing hyperbole, I'd go with the 10 sec readings and say my turntable has perfect speed.

With a heavy record, the speed was fast by a mere 0.02%.  This is very accurate, 1/300 of a semitone.  A review of the very latest Technics SP-10R shows a long term speed error of 0.12%, about six times larger (unless I'm misunderstanding this).  I'm not even sure if the Feickert Adjust record or the PlatterSpeed app are 0.02% accurate.  I'm worried that if I tried to readjust the motor screws myself I might never get it that accurate again.  It's hard to get any kind of adjustment that close.  

Update: Specs for the SP 10 Mk3 include less than 0.001% speed drift.  But perhaps you still have to set the speed, since it's variable?  If there's only a speed knob, it might be difficult to set to 0.02% in the first place, as on many table with variable speed.  Speed drift for the Sony PS-X800 is 0.002% and it has a "quartz lock" button you can press, so there's no need to dial in the correct speed first.  If the table is currently working, as mine isn't.  So perhaps 0.02% is not so above and beyond.  Or maybe "speed drift" isn't deviation from actual speed, but change, and I don't see the Linn ever changing either--you can play records all night and it's just not going to change, I believe*, perhaps it has "speed drift" even less than 0.001%.  Curiously specs for average speed, or whatever it is I'm measuring here, are hard to find generally.  (*The measurements here were taken at significantly different warmup times, on two different nights, and still seem to fit a weight curve that seems plausible.  I grasped that meant stable speed, but it wasn't very good proof or measurement.)

The Heavy record was Broken from Nine Inch Nails with a fake "blank" side.  I didn't want to risk any of my actual 180g records, which are probably slightly heavier and might therefore play even more accurately.

With a typical medium weight record, the speed was fast by 0.06%.  That's 1/100 of a semitone.  The speed with the Feickert Adjust record by itself aren't important because no 12" records will be that light, and it's still better than the Technics.

If the speed has shifted from 2 years ago, it hasn't shifted much.  Since I didn't measure it with this method, I can't really tell if it has changed at all.  Maybe it's exactly the same as two years ago, or maybe it's drifted toward even more accurate speed.

I do now have an explanation for why, even if accurately assessed, the center wow might go slightly more in one direction than another.  The center wow interacts with the cartridge-tonearm horizontal resonance, which could be zigging on one side of the record and zagging on the other, as well as constantly fluctuating.  Further influences include the damping in the cartridge itself, the anti-skating mechanism, and the levelness of the turntable.

The curious design and construction of the Linn Sondek LP12 has drawn ridicule from my friends for decades.  And I used to be of that kind of opinion myself, though my opinions evolved until I actually bought one in 1998.  Since about then, I've believed that what often appears as cheapness or madness in the original LP12 is really cleverness.  Essentially the design relies on acoustical interface principles to reduce noise transmission from the motor, bearing, and acoustical feedback.  For example, the slightly flexible subchassis is designed to reject the bearing noise caused by the single point bearing design.  It was "tuned" for that purpose.  Only when Linn developed an even quieter bearing design was it possible to replace it with a stiffer subchassis.  Even the tiny rubber feet on the bottom of the plinth are tuned to a different frequency than anything else to keep external vibration from entering the turntable.

The hyperbole official (listen to the tune) and unofficial (pace and rhythm) obscures the actual fundamental design principle behind the LP12, which comes from basic audio engineering.  Reduce noise.  

The Linn is lower noise than the seemingly more solid original AR turntable, and even the almost identical seeming--but heavier built--Ariston RD11.  I see the LP12 as a tweaked RD11.  The LP12 has a single point bearing, and a subchassis designed to reject the noise that bearing makes.  Those were refinements added to the RD11, which was otherwise a very good design, to reduce noise even more.  And since then, of course, there have been endless more upgrades.

Though, I add my own principle to that when possible (as to the Ittok XVII tonearm, which I've covered in grams of hockey tape).  Reduce resonance.

Revelations of Arm/Cartridge resonance features

I recorded the second side of the Hifi News Test LP (HFN 001) on both my Linn Sondek LP12 Valhalla with Ittok XV II arm, and my Mitsubishi LT-3 linear tracking turntable.  Both have Dynavector 17D3 cartridges and used seperate Emotiva XSP-1 Gen 2 preamps.  Both arms have been covered with hockey tape to add grams of equivalent mass to the arms, but I went further with this process on the Linn and added a mountain of hockey tape to the headshell.

Needle Drop on Linn

Needle Drop on Mitsubishi

After the need drop and before the first tracking tone, both turntables show a region of driven LF resonance.

This resonance has an envelope which has a cycle time of 1.8 seconds, which is 60sec / 33.33.

The envelope is significantly greater magnitude on the Mitsubishi, and the resonance(s) within the envelope are higher frequency.  Both undesireable.

All consistent with higher equivalent mass, AND somehow more damping, especially lateral damping, on the Linn.  This results in lower resonance added to playback, which is desirable.  (I realize there many Linn users follow an anti-damping philosophy, but I do not.  I think damping is generally a great way to control undesired resonances.)

My theory has long been the anti-skate mechanism on the Linn adds damping.  I did some tests last year but made no pictures of that.  Otherwise, the Ittok seems to have no damping.  Adding damping through the anti-skate mechanism is a brilliant idea, IMO.

The too-low mass on the Mitsubishi I can fix with more tape, I think.  I had to augment the Mitsubishi counterweight already.  I found an M6 bolt which exactly fit the hole in the back of the arm.  I think I have some more range now than when I quit adding tape, but I could always get a longer bolt.

The damping, well I don't know.  That's at the point I should try to get another turntable working rather than mess too much with the Mitsubishi linear arm.

The cyclic envelope of the resonance on both tables is 90 degrees from it's beginning when the tone burst starts.  That shows it is tied to something on the record.

I wondered for awhile how it could be that the record imposed a cyclic variation, but there was no visible square wave or anything showing where that cycle was being imposed.  Then it occurred to me:

The center hole offset causes the arm to move back and forth from ideal position, synchronized with position on the record.  But the movement imposed by the center hole offset is sinusoidal.  It only gradually reaches the peak movement in either direction.  Nevertheless, there is a distinct point on both sides of the 1.8 sec cycle where the direction reverses.  Even though that position is approached very gradually, the reversal imparts a force driving the next cycle of resonance, and especially on the inside, where it opposes the general direction of arm movement.