Update on Kenwood 600T

I'm finally about to remove the Kenwood 600T from my kitchen audio system, where the tuner is by the side of my chair and so I do the most "tuning" of different stations.  I had only moved the 600T into the kitchen to test my antennas, but ended up keeping it there for 5 years because I never got around to testing the antennas before that.  It sounds fairly decent (never have hopes THAT high for FM anymore anyway) when the Mode is switched to Filter.  That rolls off the highs just enough to remove the high frequency brightness and glare that is sadly characteristic of Kenwood tuners until they got into analog multiplier MPX circuits, as with the most famous L-02T, and the later lesser known but still stellar unit I have, the L-1000T.

I had sold my KT 917 in the 1980's way too cheap because I never liked that sound.  Much later I hoped the 600T would be sonically nicer but it wasn't.  They are somewhat similar in concept and build but differ in circuit details with the  KT-917 being much more famous.  I think the KT 917 also had a filter position, as well as a separate blend control, but I didn't pay attention to those things then.

Anyway, I don't think either tuner has the there-ness as many other high end tuners because the primitive Pulse Count Detector used by both 600T and KT-917 is fundamentally inadequate, something Kenwood themselves were aware of and fixed the problem in the L-02T and later high end tuners, with their PLL Detector, aka "linear" detector.  This should not be confused with PLL multiplex, which was already a common feature in the late 1970's.  Basically all tuners with chip MPX have PLL multiplex, and that is not even such a big deal as getting the detector right.  Before the PLL detectors, which had a slow roll out in the most high end models of the 1980's, most tuners used old fashioned methods such as ratio and quadrature detectors, which have known non-linearities.

Anyway, the slight lack of ultimate transparency in the Pulse Count Detector is of little consequence in practice, since few take FM broadcasting seriously enough to utilize the full potential of the carrier.  Later similar pulse counting detectors were made with much higher resolution (such as the Accuphase tuners of the 1990's and beyond).  Interesting that Accuphase itself was created by former Kenwood engineers in the mid 1970's, and their original tuners which competed with the top Kenwood Pulse Count Detector models used conventional detectors.  It's almost as if they left because they felt it wasn't any good, and they wanted to prove it.

But we'll see how I feel after I switch in some other tuners, notably a Pioneer which was one of their "greats" from the late 1970's, like the Pioneer ST-9500 MkII.  I think it has a commonly used quadrature detector, well perfected by that point.  I know when I used the Kenwood L-1000T there was always an incredible transparency which no other tuner I'd tried possessed, it will be interesting to see how the 9500 MkII compares to it.

Anyway, I'd long denigrated the DX'ing of both the KT-917 and 600T.  I've described the 3 position IF bandwidth switch (which as labeled as Narrow, Normal, and Wide) as "Wide, Wider, and Widest."

But now I can say with absolute certainly that while the Narrow on the 600T is nothing at all like the Super Narrow on the McIntosh MR 78, it still has some use.  I've found several stations, typically low power stations not that far away, that can simply not be tuned in until you select the Narrow IF band.  (The narrow band lets the tuner be more selective, by blocking stuff on either side better.  It doesn't help with capture issues, however, for which Wide may work better.)

Stations are still listenable nearly down to 10dBf (on the meter) which is pretty amazing.

An attached scope is the best way to adjust tuning and/or antenna.  The multipath meter is nearly useless.  The signal strength meter can be used in tuning only if you very carefully dial in the very highest peak.  Around the peak for quite aways it changes very little so you have to look for very tiny deflections of the meter.  That correlates exactly with the picture on the scope, as expected the now quite out-of-alignment center tune meter does not, though it seems like on every tuner I've ever used, aging has caused the correct tune position to be at the far right of the central mark in the center tune meter.  This is usually just before the stereo light goes out for having tuned too much higher.

My two FM Antennas Compared

 

I have two outdoor FM antennas that were installed by electricians 8 years ago.*  One is normally connected to the Living Room FM tuner and the other is normally connected to my kitchen FM tuner.  For almost 20 years I've kept the Living Room tuner set to 88.3 KPAC for background music listening, and I use the kitchen tuner (conveniently next to my kitchen chair) for searching for and tuning in other stations, especially 90.1 KSYM and 91.7 KRTU, two college radio stations.

(*This was a very expensive project because of the grounding requirements.  Each antenna has its own grounding rod, and all grounding rods are connected to the main house grounding rod by a thick wire that runs 3 feet underground across the back yard.)

Five years ago I decided I wanted to measure and compare the two antennas on the same tuner.  So for this purpose I moved my classic Kenwood 600T FM tuner to the kitchen.  I was going to use the 600T's marvelous 10dB calibrated signal strength meters, multipath meter, and scope outputs with a scope to compare my two FM antennas.  Otherwise, the 600T is not my favorite sounding tuner.  I planned to do the test in a few days and then replace the 600T with one of my other tuners that sounds better.

But after setting it up in the kitchen, I discovered the 600T sounds fairly good if the MPX Filter position on the Mode control is selected.  (Otherwise it sounds bright and tinny.)   And because of that and other things, you know how it goes, I never got around to doing the tests until now (last week).

One antenna is the famous Magnum Dynalab ST-2.  Another is the less well known Godar FM DXR 1000.  They are both whip antennas not because that is the best but because that is the easiest to install, since they can simply be screwed to the side of the house and not atop a mast.  I've long dreamed of having an antenna mast but it would be a very complicated outdoor project, too complicated for me so far.  Whip antennas also have an advantage in that they are omnidirectional and never need to be rotated.  This can also be a disadvantage if you want to block some strong unwanted signal.

(Most of the differences here probably have little to do with the whip antennas themselves, which should have fairly comparable performance in the FM band, but in the differing heights of the antennas as installed, and the fact that one of them--the Godar--is now bent.  Strangely, however, the bent antenna actually worked better on some stations in this test.)

I have the ST-2 is mounted as near as it could get to the peak of the roof, as high as I can go just mounting the antenna to the side of the house.  I couldn't mount it right at the peak because it would interfere with my locked attic door.

The Godar is mounted about 4 feet away, and situated about two feet lower because of the roof slope.  The Godar is tuned to what I determined to be best by SWR in the low FM band, probably 88.3.  The ST-2 has similar fixed tuning, best at the low FM band (which is where all my stations of interest are anyway).

Sadly tree branches messed with the Godar last year, and is bent to an angle less than half way up.  So how well is the Godar still working anyway, I wanted to know.  Do I need to buy a new antenna?

The Godar antenna, because it has greater RF bandwidth* than the ST-2 is also used for my Uniden SDS200 scanner--whose importance I'm beginning to question now that police bands here are 100% encrypted, which Scanners can't legally break even if they could do it technically.  Supposedly I can still get fire and EMS communications though I need to update the programming in my scanner for the newest systems, which changed about 2 years after I bought my scanner.

*I long figured the Godar has greater specified RF bandwidth, up to 800 Mhz because of having a lower inductance choke.  However it might also be because of the thicker whip made out of aluminum rather than steel, and because you can make the antenna much shorter.  I have it extended to nearly the maximum length so maybe I am not getting the top RF bandwidth after all, but it did work OK with my scanner at frequencies around 850 Mhz when the scanner was still programmed correctly for local services.

So in between the Godar and the living room FM tuner there is a splitter, which reduces the signal sent to the FM tuner by 3dB.  I tried an FM/TV splitter, which could theoretically have zero reduction in the FM band, but decided that it curiously caused non-linearities in the FM band which were even worse than a splitter (or so it seemed).  So I went back to a plain vanilla nice quality 1000 Mhz bandwidth splitter.  The signal then passes through permanently wired RG-6 panels in kitchen and living room to reach my living room tuner, a Pioneer F-26, one of the best FM tuners ever (like many of the other FM tuners I have collected) and from the late 1970's.

I was motivated to do the test now both because of the Godar antenna being bent and because I really really want to start playing my other tuners, including the Pioneer ST-9500 MkII that I finally got down from the top shelf in my climate controlled storage building a few months ago.  A friend believes that to be one of the best tuners ever, possibly even better than the F-26.

It just happens that I haven't been listening to KRTU much on the kitchen tuner recently, but I continue listening to 2 hours of weekly space music on KSYM.

Now finally testing the Godar on the 600T playing KSYM, I was shocked to find that the signal level was 13 dB lower, and not inconsequentially (26dB vs 39dB).  But over time I became aware that it was actually sounding better.  This became more and more clear as listening went on for a few hours.

At first I was thinking the noisier sound with the ST-2 was because of some kind of distortion, possibly caused by ground loop isolator.  Over the years, I'd used a number of ground loop isolators with the kitchen tuner because otherwise I'd get a serious hum on the kitchen audio system.  Some of those isolators seemed to cause notable distortion.  But it appears that I've removed all the isolators, and now I simply run the ST-2 signal through the RF surge suppressor built into a Monster brand AV power strip.  This also ensures it is firmly grounded to the kitchen audio system ground (or vice versa).  For a moment I was thinking of bypassing the Monster surge protection.  But it would be a big hassle because of where that power strip is located.

So what else could be causing this problem that makes the ST-2 sound worse on KSYM, I thought to myself, despite having much higher signal strength.  The Multipath meter hardly ever even moves, so it didn't look like it would be "multipath."

It was necessary to hook up the scope to find out.  And it turns out the scope is far more revealing of "multipath" distortion than the multipath meter.  In fact, the 600T multipath meter is basically useless.  You need a scope to see what is actually going on.  

Mitch Cotter, who had done a study on FM tuners for Consumers Union, convinced Saul Marantz of this, and the result was the legendary Marantz 10B tuner, inroduced in 1962 with a built-in scope, and many successors with built-in scopes made by Marantz and other companies.  After the Marantz 10B was introduced, most top tuners from other companies, even if they didn't include an actual scope, have generally included scope output jacks that could be connected to scopes.  McIntosh top tuners had the scope outputs and McIntosh themselves made a purpose built scope for consumers, whose last incarnation was the Maximum Performance Indicator MPI-4 (leave it to McIntosh to give it such a fancy name).  Kenwood made various scopes primarily for amateur radio (I have one, it's too big and clunky to mess with much, and doesn't show a good picture except for Kenwood FM tuners) but also a few for consumer audio purposes.

I've normally used an aging and increasingly dysfunctional Tektronix CRT scope for this purpose, but this time, for the first time ever for me, I used my relatively easily moved and situated Rigol DS1102E digital oscilloscope.  I had never figured out how to use this as an X-Y vectorscope which is what is needed for seeing FM multipath.  It turns out that this is an X-Y option in the Horizontal menu for this.  

KSYM using Godar DXR 1000

Hooked up to the Godar antenna, the scope shows a notable V pattern.  The thickness of the V is no more than 2 screen divisions.   That "thickness" is what indicates the degree of "multipath" distortion (which technically means unwanted in-band signals which can be caused by many things including multipath, adjacent stations, and other in-band stations).

KSYM using Magnum Dynalab ST-2 antenna

Hooked up to the ST-2 antenna, the pattern might best be described as "blob" but more importantly the middle doesn't condense down to just one screen division.  You see a smattering of dots going nearly up to the top of the screen.  It's not much of the total pattern, but it's enough to make the FM demodulation much noisier.  The multipath meter wasn't showing the total spread, just some kind of average which loses these details.

I believe what the V pattern is indicating is the presence of an on-band but weaker signal, which is being rejected by the FM tuners "capture" characteristic, which gives the upper hand to a slightly stronger signal at all modulation levels.  This is specified in an FM tuners Capture Ratio which is very good for the 600T tuner, 0.8dB in wide mode, which is almost a miracle except that a few tuners, like the legendary Sansui TU-X1 do even better.

(And tellingly, the Wide FM band selection always sounds better on KSYM, even though there are alternate strong channels on both sides.  This alone suggests the interference is in-band.  And there are several in-band candidate stations possibly causing this in-band interference, including one at 100 miles away at a higher elevation--KTXI)

What seems to be happening for the ST-2 is that since it rises higher, it's picking up more of the offending in-band interference signals, which are coming from farther away.  So it's not so much the channel I want that's getting stronger, as shown by the signal strength meter, it's the other channel I don't want that's getting stronger even faster, and just enough to start overwhelming FM capture at the fringes, enough to add some very annoying noise.

I can just as well listen to KSYM on the living room system with the F-26/Godar for space music as background music from now on, and take advantage of the superior sound of KSYM on the Godar.  In the kitchen, where I'm usually sitting in front of computer screens, the sound can be too in-your-face for background music purposes, especially during the announcements.

And it turns out that the Godar is also working better on KPAC as well, the station I usually use it for too.

On KPAC the Godar measures about 10dB weaker (only 3dB probably caused by the splitter, 7dB caused by the lower height antenna) but the multipath trace is about half the thickness of the ST-2.  Despite the lower signal strength, the sound is just a bit cleaner with the Godar antenna, as suggested by the mulitpath traces on scope.

KPAC using Godar DXR 1000

KPAC using Magnum Dynalab ST-2

Both antennas have the V outline similar to the Godar antenna on KSYM.  I think this is another capture situation with the higher antenna picking up more of an interfering station.  However, not as bad, it doesn't have random dots reaching to the top of the screen.  The main strong signal (at about 60dB on the better sounding Godar as compared with 72dB on the ST-2) is sufficient to suppress the interfering stations much better, even though it's 10dB weaker.  So the overall situation is similar to KSYM on the Godar antenna, just somewhat less so.  The meter says it's worse but the scope and actual results are that it is better.  But the sound and visible multipath on the scope are very good with either antenna, it's hard to tell the difference.

I see it may actually help for KSYM and KPAC that the Godar antenna is now bent, since this might add some directivity, suppressing other directions.  Fortuitously, it's bent in the direction of the transmitters of KSYM and KPAC.  The bend would tend to block stations coming from the sides FWIW.  Since it's working so well I don't need or want to touch it.  Perhaps a brand new antenna would be higher and therefore worse just like the ST-2.  If the Godar finally breaks off I'll probably have to replace it then.

So it appears that both the signal strength and the "multipath" meters are useless in situations like this, and probably many others.  You simply need to have a scope to get a feel of what's going on.

The best FM antenna is a big multielement antenna on a rotor on a mast, which has long been my dream, but I've hardly been able to contemplate setting it up, and basically nobody does that kind of thing for hire (at any price a mortal can afford, anyway).  I called "antenna installers" and all they wanted to do was tack up a small VHF antenna for digital TV to the side of the house, just as I have done with the whip antennas.  (I have two such brand new unused antennas in my junkpile, a legendary APS-10, and a classic Radio Shack FM-6.  How to put em up has always been the problem, as well as the grounding stuff I got sorted out for my whips.)

I also dug up a quick RF disconnect for one of my antenna cables and if I install another one on the other cable, I can swap antenna cables as needed (if I want to listen to KSYM on the kitchen tuner, for example).

KRTU still measures and sounds better on the ST-2, and by no small margin.  It has 10dB more signal strength AND less "mulitpath" visible on the scope.  Even less than on KPAC with the Godar.  It's obviously best to listen to KRTU on the kitchen tuner with the ST-2, which is what I always do.

KRTU on Godar DXR 1000

Despite having less apparent in-band interference, the multipath display for KRTU on the ST-2 still shows the V shape--and perhaps even finer than the stronger KPAC.  So I tuned around the dial a bit, and it appears all the stronger stations show that "V" characteristic, even with signals as high as 75dB.  So it appears that the V shape is just what the 600T tends to show on its scope outputs, given the way it's designed to produced the scope outputs.  That actually varies a bit from one tuner to another.  I would have expected the ideal display to be a flat line.  But "idealness" for the 600T appears to be how thick the trace of the V is, or if you can't even see the V at all because it's so blurry.  It's also great for tuning as the tiniest off-tuning results in a detectable tilt one way or the other at the bottom, which is flat right at the center.  The meters are more difficult to read or interpret.  For precise fine tuning, the center tune meter is fairly useless as it's correctly tuned at the far right of the inner bubble (a through tuner alignment would fix this).  The signal strength meter is not biased like that, but around the center tuning position there are only very small changes.

"Scoping" cables?

 A friend wants to know if it would be useful to examine high end cables on one of my oscilloscopes.  I tell him my equipment is inadequate for such purposes, it would be much better to have a 500 mHz or 1 gHz scope.  He incredulously asks me why such is needed for audio frequencies.  It's hard to answer that question in a phone text, so I'm doing it here.

There are a number of different aspects to this, so I may seem to go all over the place.

1.  Macro vs Micro parameters

Oscilloscopes and voltmeters measure what we could call macro parameters of audio transmission.  Such parameters could include DC voltage and risetime.  Basically things having to do with the large outline of the signal transmission: bandwidth and phase response.

Micro parameters like noise and distortion are not so easily measured with oscilloscopes and voltmeters.  Best thing to do is store the signal and apply an algorithm to compute them, such as the program RMAA (but you'd probably need something much better for audio cables...*)

Noise can come from many sources: internally generated and picked up.  How well audio cables accept or reject noise from EMI and RFI is in fact very important but often ignored.

2.  How do audio cables differ?

Decent audio cables don't generally differ that much in their macro parameters.  The bandwidth of a typical RCA terminated coax cable is around 10 Mhz or greater.  Generally you can use such cables to transmit composite video which has sidebands up to 6 Mhz (though good video cables will often have a bandwidth 10-100 times greater than that for the best performance).  I have often used regular audio cables for both video and digital audio and it nearly always has worked (though I generally like to get the proper cables, etc).

It's not surprising, therefore, that many engineers and audio scientists believe "audiophile" cables are not necessary.  There's no way, they might say, that you need more than the 10 Mhz bandwidth of typical interconnects.

But they aren't generally considering micro parameters.

3.  But I've seen scope comparisons of cables and they showed differences.  How is that?

I've seen such photographs too, though I can't remember the exact details, I remember once when I really checked it out they were using a very high bandwidth scope to resolve differences at very small timescales.  That means high bandwidths, and they were using 500 mHz or 1 gHz scopes.

Now it should be remembered that you always need about 10 times the scope bandwidth of the signal you are trying to view.  If you view a 10 mHz square wave on a 10 mHz scope you may see a sine wave attenuated by about -3dB, because that's how bandwidth is defined.   You see none of the harmonics that make the wave square.

The square sides of a square wave are created by a series of harmonics which in principle go up to infinity.  That's why they needed very high bandwidth scopes to see details of very small duration (representing high frequency effects) in "audio frequency" squarewaves. 

4.  But I'm sure I don't always get 10 mHz bandwidth.  I can hear the rolloff.

Yes, but such issues likely related to the input and output impedances being applied, in combination with either the capacitance (for an interconnect) or the inductance (for a speaker cable).  These macro parameters can serve as a low pass filter.

That isn't really a fault of the cable, it's a mismatched system.  Now audio cables are too short to show what are called Transmission Line Effects at anything like audio frequencies.   Such effects can be shown for audio cables running many miles, but not a few feet.

So audio cables are operating as a lumped sum system.  You can easily calculate what its response would be if you could accurately know the output impedance, input impedance, and cable capacitance and/or inductance.

So to "match" cables, for example, you would select two having identical capacitance or inductance.

Just by doing that, you've matched the macro parameters.  No need to use a scope.

You can do that on a scope, too, but what you are showing, strictly speaking, is not the performance of the cable itself but of the system with input and output impedances.  It would have be measured connected to the intended input and output equipment of interest.

I've measured many 'plain vanilla' interconnect cables and they tend to have between 11 and 60 pF of capacitance per foot, with 30 pF being typical.  That is not going to have any issues for most audio equipment, but sometimes audiophiles have equipment with unusually high output impedances or unusually high input impedances (in some tube equipment for example) where there might be issues.

With speaker cables, the impedances are all more critical, and I don't have any "inductance" meters.  A scope would be the way to see the macro parameters.  But with reasonably well matched impedances, you aren't like to see any differences among cables at audio frequencies.  The way that speaker cables interact with amplifier and speaker loads in through very tiny effects.  Say, the impedance may be 0.05 ohms at one frequency and 0.04 at some other frequency.  Against an 8 ohm load, that would cause a very subtle change in timbre, but not something you are going to easily see on a scope.  It's a miniscule, but potentially audible effect.

*5.  You're not trying to tell me you don't believe in cable differences?

I'm trying to remain open minded and non-judgemental as to what you or other people can hear, in writing this post.

Frankly I don't believe there are significant audible differences among decent audio cables.  You could probably get by with plain vanilla radio shack cables in most cases (and I still have boxes of them**).

If there are significant differences, they are most certain to be found among the micro parameters and not the macro parameters, because in the macro parameters such differences just aren't there at audio frequencies, you can't measure any such differences easily, which is why many engineers and audio scientists believe that decent cables are the same.

**Nowadays I try to go with what is objectively best anyway, just in case it might be important.  So I have PTFE coated speaker wires because PTFE has the lowest Dielectric Absortion of all solid insulating materials, and a number of other considerations.  I have also chosen my speaker cable gauge (according to some principles that are too complicated to explain here, but end up making 12g the optimal for twisted pair) deliberately. But are such things necessary?  I doubt it.  For most interconnects, I use BJC-1 from Blue Jeans because it has the lowest capacitance and best shielding.  It also uses polyethylene dielectric, which is nearly as good as PTFE, but that might not even be important.

Dielectric absorption, in which an insulator operates as a battery, is largely a low frequency and low level phenomenon, something you are generally not going to see on a scope unless you test then with DC levels and very high resolution measurements.  There is very little difference among the 3 currently popular materials: PTFE, FEP, and polyethylene because they are all so good.  They are largely unaffected by electromagnetic fields at the molecular level.  Vinyl is a crappy dielectric, but probably adequate for audio purposes, I used vinyl speaker cables until around 15 years ago.  

But one of the worst things about old speaker cables is copper wire corrosion, which most frequently occurs with vinyl cables.  Sufficient corrosion can cause distortion, potentially audible, but again not measurable on a scope.  (And maybe you'd even like the effect if you liked a "fuller" sound.)

6.  Haven't you heard cable differences?  Here I can show you right now...

Yes, I have sometimes heard 'differences.'  However not under the circumstances that I would consider made it a valid experiment.

A valid audio experiment includes these details:

1) Level matching of all sources (etc) to 0.1dB.  You must have perfectly repeatable audio levels, such as with digital controls or stepped attenuators, rather than continuously variable controls.  Tiny level differences (as low as 0.25dB in my experience) can change the quality of a sound without changing its apparent loudness.

2) Listening position and environment must not change.

3) Double Blinded testing should be done, where neither the experimenter nor the subject knows which is which, along with a suitable protocol such as ABX.  This is especially true if there is any amount of time passes between one audition and the next, because it is relying on a fading memory for comparison.  Repeat trials are necessary until the null hypothesis can be rejected all but one times in twenty (p < 0.05).

Without these controls, I don't consider any listening test to be definitive, but just a guess.  Peter Montcrief created the term Golden Eared Subjective Reviewer (acronym GESR) to describe many well known audiophile reviewers (not that he didn't have his own faults).

Listening tests done without such controls can be fun if people don't take them very seriously, as they should not.

Music Distribution Services

I was elated to be able to finish two new music albums and upload them for distribution in October.  First time since 2022.

But now it appears that the "free" distributor Routenote has disapproved the first album I uploaded without any messages to me.  The album simply disappeared about 1 week after submission, without any email, explanation, or anything.  You have to ask what went wrong now.  I think it's likely the second will meet a similar fate, as it's not any better.

Routenote has been successfully distributing my 'Mythic Rocks' album, though the earnings so far ($1.36 on 1500 plays) are pitiful.

Pitiful earnings are likely anywhere, but I need to go ahead with some kind of distribution, because my music is not something I can play at street corners or bars.  I can't 'develop' without some kind of audience.

A paid distributor like Tunecore or Anti-Joy might work better.  Routenote has become notorious for their unhelpful rejections.  I released my very first album on Tunecore in 2006 but quit paying for distribution long ago, thinking I could do just as well with free distribution.  Free distribution worked OK for Mythic Rocks anyway.  (It didn't work with a second album I tried to release in 2022 because I was recycling some of my older songs I had previously distributed as if they were new songs, but under a new artist name, and I should have known Routenote would figure that out even if the song was no longer being distributed the metadata persists.  But I didn't make that mistake this time, everything was all new in the albums I attempted to release in 2024.)

Now it looks like the charges at Tunecore are reasonable and anti-Joy might be even better.

Here's a (slightly outdated) survey from about a year ago.  The author mentions 10 services and doesn't even mention Routenote (which figures I guess, they've become so notorious--not just for me).

Here's another survey with very colorful user reports (many are fed up with most or all distributors).  Routenote does come up with one poster initially saying it's great but then calling it awful after all 10 of his submissions were rejected.

Here's another 'complete' review of "all" the services, but lacking the colorful comments by users about their experiences.  They only put down Routenote as not having the most complete service (and nothing about its infamous review process, "customer service," etc).  This is a useful resource*, but limited, and it seems a few services highly praised in the earlier reports are missing (notably anti-Joy, which seems to be the favorite at the first link I posted).  Actually, what we really want to know, is which distributors are assholes.  Personal reviews call them out, though problematically every service is called asshole by someone.  The linked detailed reports are more useful and may give some personal experience, but limited to that of the author who is an accomplished musician and knows how to deal with these people.

It appears now that a submission to Routenote is more like an audition, where you get a phone call three weeks later if you've been chosen.  They aren't a 'distribution service' but more like a record producer (in fact, they are a record producer).  The want the reputation among streaming services as a consistent provider of Good Music, which has it's perks like getting to the front of the line.  They also don't want to bother with music that isn't likely to pay them sufficiently in commissions to justify the costs.  IOW, they're likely looking for a set of albums to gross around $75/year or more of which 15% would be theirs, or about $10.  From an artist who hasn't yet, but just might break through, which they could see a month of or more.  Though they accepted my Mythic Rocks album when I submitted it, and it's only paid about $1.36 in two years, of which they have pocketed everything for now because it doesn't meet their $50 or $100 minimum, they may have fine tuned their algorithms since then to reject such freaky stuff in the future, because it's not just worth the costs for them.  And then the fear of bots, etc.

One musician described Routenote's opaquely applied standards as Fascism.

Routenote has never been the top distributor, and often they aren't even included at first in lists.  The 'free' model got them a lot of attention at first but it's all cooled now as people find out that means they're very picky, slow, and opaque, and people who are good enough to pass muster can probably afford a better service with faster updates, more info, more customer service, etc.  I only found them because they advertised a high definition capability, and I figured that would mean I could upload 'high definition' files.  But, as it turned out, you have to be a major artist for that capability to even be available.  I have always sent, and they accepted, 24 bit flac files, but streamers will only see 16 bits because that's the way the system works, High Definition is a "premium" for "top albums," everyone else gets 16 bits.

If they'd give me a list of what's actually wrong that would be helpful.

While I originally thought the commission model would give them some incentive to help me along, in the final analysis the opposite is true.  The commissions they expect to make from me aren't worth their efforts at all, they figure, so no I get help at all, as I might at least get modest help from a company who at least hopes to keep pocketing my annual fees.

It appears I'm just going to have to go back to a paid service.  I'm leaning towards Tunecore again.  Everything is compared to Tunecore, the pioneer, and they've now re-designed their current pricing tiers to do better than major upstart alternatives.  Annual fees cover unlimited albums, and most authorities say you have to release about one a month to make money now.  They're widely reported to at least have reasonable customer service.

*The overall review section is useful, but devoid of actual experiences, and therefore reminds me of the reviews I used to read in Stereo Review magazine in the 1970's.  Everything is good, but some things have more good features than others.  That sort of industry-submissive attitude was undermined by audiophiles who wanted more dirt, and Stereophile and The Absolute Sound were born and Stereo Review went bust because nobody wanted to bother with their pablum.  But Audio magazine was a bit more informative than Stereo Review and Audio went bust anyway.  Actually I think Stereophile and The Absolute Sound were and still are too cranky, but sometimes fun and I don't miss Stereo Review.

Optimal Surround locations

 For Dolby Atmos, it's 1 foot above and 6" in front of ear, according to a UK magazine:

https://www.audioadvice.com/blogs/expert-advice/home-theater-speaker-layout-options

That's a bit less than 90 degrees.

Google AI suggested 90-110 degrees from center from some other source for both 5.1 and 7.1 "surround."  That's what official Dolby Atmos website shows for 7.1.  But for 5.1 Dolby Atmos shows 110-120 degrees in their diagram but the picture shows 90 degree direct side placement.

Crutchfield says "to the side pointed at listener" for 5.1, without giving any specific number of degrees, but the picture shows the 5.1 surrounds directly to the side, showing the Dolby picture (but not their diagram).

Discussion at AudioScienceReview quoted ITU-R international standard which specifies 100-120 degrees.  They also clipped a bit from Floyd Toole's book showing arguing as you get closer to 90 and 120 the effect diminishes.  (So, in this telling, 90 and 120 are equivalent, so it seems to me the range should be 100-110 or 90-120.)

But Floyd Toole doesn't actually show 100 or 110, just 90 and 120.  And the 90 and 120 graphs (which are not entirely comparable, as the 90 degree one has no center channel) have virtually identical scores!  

Toole also shows a surround setup with speakers at 30 degrees and 60 degrees, and that seems to have the best score of all!  But I haven't seen any discussion of that.  Yamaha famously had a 7.1 or 9.1 system with wide front Presence speakers as one option.  I have one of those receivers still in my Kitchen which I use mostly for stereo but will do some semblance of 5.1 and 7.1 either direct from Oppo or synthesized by receiver with my own manual fine tuning.  I have never done wide front 'presences speakers, though my front speakers are 30 degrees (I myself prefer 45 degrees as in my living room system for the widest possible stereo, which also means the most information).

Here's what leading Harman scientist (under Floyd Toole) Dr Sean Olive said about surround layout (in 2018):

 For example, if you’re setting up a stereo system around a TV or project screen, you’ll want to try and get the right and left front speakers flanking the screen at ±30 degrees angles relative to your seat in front of the screen to ensure a best-class sound experience. For surround sound the left and right side and rear speakers should be at angles of ±60-100 degrees, and ±135-150 degrees, respectively. After setting up the surround sound, it’s critical that consumers leverage the test sequence in the receiver, which will help ensure that every speaker is set at the proper level and time arrival.

Perahps Harman was not on the same page as Dolby and others.  The Harman ideas were based on their own objective psychoacoustic research in their laboratory in California.  (However, I need to get more detail on Dr. Olive's reasonings. than provided here.)  Perhaps it's in Toole's book, which I think I have.

But also note that Dr Olive is implicitly assuming both side and rear speakers!  So it's not really answering the question here (what's best for 5.1) but 7.1, and for 7.1 nearly all authorities either include 90 degrees in their rage (often showing it in pictures) or specify it exactly.  But for a room of listeners, only those lucky enough to be on axis with the side speakers will get 90 degrees, and those in the back will get...well 60 degrees if they're lucky (a point seldom discussed).

Random commenter at AVSForum suggested 90 degrees for 7.1, and somewhere in between that and rear for 5.1. 

I could easily do 90 degrees in my living room for listeners on the back couch which is right on the back wall.

Some don't think that is right even for 7.1.

Then what are you supposed to do for couch on back wall?  Every other side position is further front.

"New" sub polarity is non-inverting

 I strongly believe now that the reversal in sub polarities I did during the Janus 1.0 system tuning was correcting a number of earlier mistakes, but the polarity of the subs is now non-inverted, which is what I intended them to be.

I most recently replaced the plate amplifier in the right sub.  That was a couple years ago, and I decided I'd do it right this time for sure.  So before hooking up the plate amplifier, I connected an extension wire to the sub driver so I could stand in front of it and watch the woofer go in or out as I connected a battery to it.  I had never done that test before.  Once the plate amplifier is connected, you can no longer do such an easy test.  If you try to use a polarity matching app as I have done, you quickly find it is not reliable and even probably wrong at subwoofer frequencies.  This is because room reflections become part of the mix before you've even had one "cycle" at low frequencies.

That right sub has a polarity menu item, and it is now set to "Normal" (it had been "Inverted" for some unknown previous amount of time, perhaps since I did the plate amplifier replacement 2 years ago).  Right now I don't remember why I set it to Inverted or perhaps it was a mistake.

The left sub, OTOH, has its polarity inverted because (so it seems) I installed the replacement plate amplifier incorrectly.  I knew this was the case the first time I replaced the plate amplifier, so I ordered a polarity reversing XLR cable to connect it with, rather than having to take the plate amplifier out all over again (which requires moving a lot of stuff around).  That was about 12 years ago.  I used the polarity reversing cable on the left for awhile, but at some point (while I was still using Behringer DCX 2496 as crossovers, and they have polarity control) I decided on a trick: I'd reverse the polarity on the left and then reverse the polarity to both subs, making them both correct.

In 2013 I started using Behringer DEQ 2496 as crossovers (because they have a digital output option, allowing me to use high end DACs) but they do not have any polarity control (though it seems now that older ones are polarity inverting), and then in 2021 I started using miniDSP's as crossovers.  Somehow in all these changes, I lost track of the fact that the left sub polarity needed to be inverted (as with the inverting cable) to be in non-inverting polarity.  I also replaced the left sub plate amp a second time, but I think I may have made the same mistake as the first time.

The problem I faced recently had nothing to do with absolute polarity (which, for subs, I now think isn't even an issue at all) but with the relative polarity between the panels and the subs.

With the 24db/octave filter, the top or bottom polarities can be reversed without having any effect on the frequency response.  Linkwitz pointed out that the "inverted" phase connection with the 24dB/octave filter has the lowest group delay, but I've always believed in having everything in non-inverted polarity simply because it's possible and more intuitively nicer.

(Although it's possible that at some point I decided to use the inverted sub polarity instead.  It should have had no effect on frequency response.)

I was thinking he 48dB/octave Phase Linear Linkwitz-Riley crossover I am using now has that same property, you can connect the drivers either way and get the same frequency response.  However, I now know that is not true, it makes some difference which way you do it and you get some cancellation at the crossover frequency doing it incorrectly.  So it actually seems that the "Phase Linear" FIR filters I am using don't have the same property in this regard as the normal Minimum Phase LR filters.  Perhaps it's because they don't have any phase shift at all, so the phases are either matching or not, there is no in-between.

Mind you, the crossover is so steep the "cancellation" (which isn't perfect either) is confined to a very narrow band around the crossover frequency, and can mostly be masked with EQ (as I had been doing without knowing it) mostly.

Janus 1.1

The stereo was sounding great on Saturday night, and I was doing some rare serious listening.  But in the morning the weird looking left channel response was bugging me, compared to the right.  And it had been much better looking, so it seemed in the last post, just before I added the last 182 Hz boost.  Then the lower midrange somehow got all messed up.  I had to solve this riddle.  And by Sunday afternoon my knees were no longer hurting from earlier days of adjusting (even using my 3" foam kneeling pad).

First I dialed out that boost at 182 Hz, which was seeming to be the culprit.  But it wasn't.  With no boost at 182 Hz it looked much worse overall, however the region 500-1000 Hz was perhaps slightly better, mysteriously.  Very slightly.  Not worth it.  Only then I did a good re-measurement of the 182 Hz boosted version for comparison, and it was mysteriously looking quite a bit better than a few days ago just by chance (not EQ), and obviously better with the boost.

Left, 182 Hz Boost removed

Left boost restored, "Janus 1.0", maybe not so bad

I tried half as much boost.  That was neither here nor there.

So then I just tried fixing the issues in the midrange while leaving the boost in place.

There was a depression centered around 400 Hz (it was a deep depression in the measurement I made a few days ago, today it was a fairly mild depression).  And there was also a -3.5dB notch at 437 Hz.  So I eliminated that notch, the the depression looked much better.  I left another -2.5dB notch at 366 Hz untouched because 300 and 400 were about the same level with the 437 Hz cut removed.

Now that I'd given up the 437 Hz notch altogether, I had an available PEQ I could use for a notch at 833, which had been used in earlier smoother iterations, but (mistakenly it seems now) given up at one point as unnecessary.  I might have intended to remove the 437 and ended up removing the 833 instead.

I moved the 903 Hz boost up to 1013 where it had been, but that left 900 depressed and added to the peak following 1khz.  So I found a happy medium where 900 and 1000kHz are about the same, tuning the boost to 957 Hz.  It's a very narrow boost so it doesn't raise the 800 Hz (I didn't change that).  For the smoothest response above and below 1kHz I increased the boost from 4dB to 5dB which gave the flattest response around 1kHz.

With that, I needed to increase the depth of the notch at 654 Hz because that was bubbling up again.  It had been precisely that I before but for some reason I'd then rolled it back.

And that was it, I got a smoothness from 500-2000 Hz which was similar if not better than the pre-182-Hz-boost response I'd measured a few days ago, but without sacrificing the 180-300 region which that boost fixed.  Actually this is about the smoothest I recall ever seeing the Left Channel.  Still not quite as smooth as the Right but pretty close.  It's now good enough that I think I can relax on the EQ'ing for awhile.

Left Channel, "Janus 1.1" EQ

Janus 1.1 PEQ's (only left panel changed)

One notch was a restoration of a notch I had figured out by sweeping and was using earlier, the notch at 833 Hz.  I basically just added that back and it helped so I'm thinking removing it was a mistake.

The boost was moved to a frequency based on the RTA effect and not sweeping this time.  That was what I actually do a lot but not what I claim to be my method because it's leaving the ultimate hinge frequency to a wider than necessary analytical technique.  But I've actually done that in many cases like this before.  Sweeping the mids and highs is not easy at all because the distances from micro peaks to micro valleys (which you would see in a very slow sweep graph, it would be "thick" because of all the ups and downs) becomes a smaller and smaller portion of an octave, and the frequencies you can choose for the PEQ's get wider and wider and there's not much you can do to get really "in there."  It is possible it might be better if I moved the boost a tick or two higher or lower based on sweeping or, gasp, sound.  But it could not get any flatter RTA, since that was being optimized.  I recall when I was sweeping that there was a dip at 903 Hz and also around 960 or so.  When you get up to 1khz, it doesn't sound like it needs boosting at all.  So the 957 may actually be just right and it's certainly pretty close to what I would have chose by sweeping.

2024 'Janus One' EQ adjustment

The basics have now basically been done for the current EQ and level (and phase this time) adjustments.  And I've given it a name to leave a milestone here, "Janus One."

It's sounding wonderful, a huge upgrade, even though I thought it was incredibly good (and basically better than anything else I've heard in a while) before.

I've eliminated the "Boost" aka "Listening Position" EQ.  Now it's called Normal and used for everything except movies--which have guests and use peripheral seating where there is maximum room gain.  Normal sounds way over the top at those positions, however audiophiles might find it ok at a meeting.

I've been able to eliminate Boost because the new EQ is so well tuned.  It isn't "sloppy" like the old Listening Position EQ, which featured boosts on top of boosts in some cases.

It's much smoother overall for many reasons.  A big part of it is the deliberate choice to NEVER use Graphic EQ's (GEQ).  GEQ's tend to increase roughness because they aren't precisely dialed in over existing depressions, and therefore often amplify peaks--and peaks amplify with gusto.

Each way and channel uses only 10 or fewer Programmable EQ's (PEQ), mostly as notches, but with a few highly tuned boosts also where they appeared to be needed.

They were all tuned using an oscillator, frequency counter, and chairside Digital Equalizer (DEQ) with all the critical listening being done in the listening chair.  Measurements were done with 1/6 octave 16k bin 'Analyzer' app, for the first time held in the center head (or sometimes ear) position with a microphone stand and new phone mount.   These are big improvements over previous tunings, where I often sat on the floor in front of the front DEQ's while tuning in the frequencies and deciding on their need for attenuation, usually using the DEQ itself to figure out what the critical frequencies were.

In the process, I discovered that it worked substantially better in both channels to invert the polarity of both subs compared to their previous positions (which in one case had been "inverted" and is now moved back to Normal).  The sub markings are in question, as I perhaps installed both replacement plate amplifiers correctly, but they somewhat agree with the ambiguous Speaker Pop app which becomes very unreliable with subs, which is no surprise because I originally set them up using Speaker Pop.

Anyway, in recap, I think it's good to go over what the change has been.

When I started, the listening position response was very bass shy UNLESS I engaged the "Boost" EQ.  That added a bunch of GEQ boosts to the normal set of PEQ cut used in all settings.  With that selected, I got a pretty rough and tumble bass response in the Right Channel:

Right Channel, Boost EQ, July 13

When I finished Janus One, there was no longer any Boost EQ as I had designed the NORMAL eq to have sufficient bass at the listening position without using any GEQ boosts (but using some very small PEQ boosts).  When I finally got the sub phase correctly set, I measured this:

Right Channel, Normal EQ, September 12

Wow!  Very smooth overall (especially from 20-400 Hz) and closely fitting what I think is the correct room curve.  No need for a Boost curve and not too bad elsewhere in room or house (just a tad too much bass elsewhere which often works like a pleasant loudness curve for background listening).  In one intermediate adjustment I had the bass completely flat with the midrange, but I decided the bass was far too weak sounding, so I raised it by increasing the sub level.

The left channel had been looking much better than the right channel, I originally thought, but by the time I'd finished the right it was clearly no match anymore.  (Now I wonder why I ever thought this was good.)

Left Channel, Boost EQ, August 1

The finished Left Channel Normal EQ isn't quite as smooth as the right is still, but much nicer for the most part.

Left Channel, Normal EQ, September 12

The bass from 20 Hz to 300 is very smooth, but above that there's some roughness.  The only way to fix 400 Hz is with a boost near there, I thought the region around 1kHz looked smoother in some previous measurements.  I try to avoid boosts and they need to be done very carefully, so not done yet.  Careful listening tests might be warranted too.  This includes "PB1" the current latest Panel EQ, which only has one boost in the left channel (but looks like it might need 2 or 3, I think now.)

Looking at it now, I think this could use some more work asap, it's still no where near as perfected as the Right channel.  I was bogged down for a week with the phasing issues so I couldn't quite see that.

While doing this EQ I learned at least one lesson for sure.  Always try inverting the phase between subs and panels, even if you think you have the correct phase.  And take a good measurement to compare them.

And many others I've noted along the way.

One part of identifying this milestone is to write down ALL the details so it's not lost later.

Details:

XLR polarity inverter on Left sub channel

Right Sub set to "Normal"

Left Sub level: -7

Right Sub level: -10

Sub DAC: -6.5

Panel DAC: +9.5

Panel DEQ Gain Compensation: +3.0

Tact Levels:

R 0

L -2.0

Panel DEQ Delays

L: 10.0 ms (design default)

R: 10.0 ms

Sub DEQ Delays

L: 8.72 ms

R: 8.40 ms

(Honestly these delays don't look small enough compared to the panels, there should be more delay differenceI think, but these were dialed in years ago after some testing and thinking then.)

NO GEQ's whatsoever!  (Except in Movie mode presently)

Left DEQ's, sub on top

I see now that there are actually 3 boosts in the left panel, and the upper frequency ones aren't doing very well.  The chairside PB1 memory is identical.  This looks like a mistake perhaps, did PB1 get overwritten by PB3 by mistake???  Looks like a week or two of rework is needed on left to banish the remaining lumpiness, not just a new boost or two.  But I've decided my knees need to rest from kneeling so I'm not going to do it now.  That's going to be "Janus 1.1".  And also asap I should start measuring with my calibrated microphone and ARTA.  I believe my iPhone 8S and Analyzer app is accurate enough for most purposes, especially when supported in my new microphone stand mount, but super fine tuning around 1kHz and above might benefit from something even better.  The beauty of using the phone analyzer is that it's so fast I can just whip off measurement after measurement, only a fraction of which have been displayed here, which is invaluable for mass EQ tuning.  You've got to test every little change because actually it's all virtually unpredictable.  (Some supercomputer with a description of every object in the room in precise detail would be the minimum requirement.)  Cut and try is the only hope.  Once you've more or less zoned in on the correct notch and boost frequencies, you can take the time for more exacting but time consuming measurements and even finer tuning.

I notice now that there are cuts near the 400 Hz depression, which I thought I had eliminated...

The better tuned Right channel does have only one boost in the panels (and it's just the low one at 133 Hz).

Right DEQ's, sub on top

The Movie presets simply add a set of GEQ's to an earlier set of PEQ's with no boosts.  (The PEQ's are not very critical, but the sub version should not have any boosts which may go awry in the far field.  It would be better to re-do the PEQ's for the back seating position from scratch so they don't need GEQ's, but that's another big project for later, heck I'm not even done with the Normal yet I realize now.

Left Movie GEQ

Right Movie GEQ

The subs also have Dynamic EQ (dEQ), a form of limiting.  It rolls off the deepest bass (to prevent overexcursion) when the level is too high.  SVS probably already has such a thing built in to their plate amplifiers to keep their speakers from blowing up, but it's not as limiting and protective as mine.  I investigated this a few years back, and determined I got the best results with two identical dEQ's in sequence, that way it can compress the maximum amount fast enough when needed.  And I eliminated the standard compression DYN because it isn't needed on the subs and I don't think it sounds good.  3 dEQ's was too much so I only use 2.  The only thing that may be out of adjustment now is the Threshold level, which depends on sub DAC and sub amp levels.  It can be re-tuned by watching the indicators when playing a bass test track.  On most recordings it should not trigger at all.

Sub Dynamic EQ page 2

Sub DEQ page 3

**** Wait!

I thought I had remembered much better looking Left channel response.  And indeed I had a stunningly good measurement on August 29 in an earlier post.  This was made using the chairside DEQ (shown in photo) so it automatically had the polarity changing adapter I added later (though I added it to the sub not the panel, the effect on response should be the same).

Left Channel, August 28

Left Panel peq's, August 28

The only difference I see here is that I subsequently added at boost at 182 Hz, replacing a 0dB filter at 824 Hz.  It's hard to believe that would cause a depression at 400 Hz, or 1kHz depressed compared with 900 Hz.  Perhaps the Chairside was not showing the EQ that was actually operating because it needed rebooting.

Running through all the published spectra and adjustments shows nothing inexplicable.  I had a mound around 1khz when I had a wide filter at 1013.  There was consistently a depression at 900 Hz before I moved the boost there instead of 1013.  Fine tuning that nearly leveled the whole region.  All looked believable, it didn't seem there was any hangup.

Stuff in the room may have been moved around a bit, but not that much I would think.  It still looks like the point where everything goes south is when the 182 Hz boost was added, which doesn't make any sense, but seems worth trying.  Tiny time delays might have been involved (I found the front panel DEQ's had 0.1ms greater latency, don't recall if that was with or without optical connections though the optical connections made no more difference than that, if at all).

Anyway, that's a wrap for the Janus 1.0 summary.  We can start to imagine Janus 1.1 now.

Re- Doing the Polarity

Now it's looking like I can get much nicer system response in both channels...with the polarity of either subs or panels (but not both) inverted from the present settings.  When either was unintentionally inverted with the chairside DEQ, I got nicer looking curves, and notably with NO dip at or around the crossover region.  With the front panel DEQ, which was not inverting, I struggled to get such nice curves and came closest when I adjusted the delay to be around 4 ms more than previously determined settings (setting which make sense and they also approximate the sub backward distance, as I couldn't see it any better than that anyway).  That 4ms is just enough to cause a polarity inversion at the 125 Hz crossover frequency.

Both were checked before in a variety of ways, including a phone app combined with its downloaded test file.  (I never trust such things very much.)

But to really know what's going on, it would be best to measure polarity myself.  And now I have an oscilloscope in the living room just ready for such a task.  I just need to hook up a microphone and microphone preamp.

I'm sure the sub output, even without equalization, will look quite different from the original polarity test signal I have been using in the past few days (either of my self produced files: SoftPolarityTest or LoudPolarityTest2, which are a series of pulses spaced by about 5ms).  It will be long and flat and even cyclic because of the rolloff and crossover of the sub.  Turning off the EQ (but not the crossover, which is not a good idea on my system) may be helpful in making the correct polarity more visible.  I might also need to record a test with longer pulses, the current ones probably don't generate much low frequencies.  My original tests were made by analog recording one of my oscillators (not the present one, which might make it easier) because I don't know of any software that does this sort of thing, not Audacity for sure, and I couldn't find any such plug-ins either.

SoftPolarityTest output from panel DEQ (including crossover)

The 4x orders of the Linkwitz-Riley filters permit both drivers to be in phase with one another.  Thus it is true for the 4th order (24dB/octave) however for that crossover, less group delay is achieved with the out-of-phase connections (I never used them anyway with 4th order, preferring to have everything in-phase just in principle).  Now that I have 8th order Linkwitz Riley, I believe not only can I use the in-phase connection, but it achieves less group delay that way as well, because that characteristic alternates with higher orders.  (IIRC.)  However, I am not sure if this is affected by the phase corrected (FIR) filters I currently use.  It was my understanding that they would simply output everything in phase all the time.  So therefore phase would not be optional--in phase would be required.

If there is some flaw in this which means I must use out-of-phase connections, I want to know that too.  Perhaps I can change the crossover to fix that.  Because the current pair of speakers lacks problems in the 80-100 Hz region, I could use a 4th order crossover again if needed (though...I think it also causes complications with wall bounce).  I could also sacrifice the phase correction thing too (if I went to 4th order, for which phase effects are still inaudible).

So the best case would be if the current sub polarity is incorrect.  It was actually somewhat ambiguous with the phone app.  Then I could just fix that.  But if it turns out that the ways work better being out-of-phase with one another, well I will do that for now but think about changing the crossover as well.

And if I do make them out-of-phase for now, which should be the in-phase part?  Opinions vary.  I don't think I could hear a difference but I suppose I should try, if that becomes necessary.  My preconceived idea is that the sub should be out-of-phase and the panel in-phase, which is the opposite of opinions I have heard from the Polarity Pundit, my friend George Louis.

The kind of solo trumbone and other asymmetrical sounds where absolute polarity has proven in some cases to be detectable mostly contain a lot of high frequencies.  In the bass, it's all a jumble of whooshing reflected sound.  The leading edge of anything, even a low frequency sine, but with acoustic harmonic sounds even more so, represents a quick blast of high frequencies because the starting and stopping of anything, it's abruptness, represents high frequencies, whereas the extension, duration, of everything represents its lowest frequencies.  The only such thing as a "pure sine" would be one that started at the beginning of time and never ends.  We normally don't think in such terms, we truncate into bits, and say "see sine wave here" and imagine that duration in isolation from the world--a world unto itself--but in the real world a sound must start and stop, and those are not actually the same as the rest.

*****

I doubt absolute polarity is reliably identified by ear.  High quality experiments published in JAES indicated otherwise with some exceptions, such as solo trumbone on headphones.  Mixed music, or virtually anything played in a room, has too much complexity for polarity effects to be audible.

But nevertheless, I do take some effort to preserve it.  It was a shock therefore to discover this week that my long trusted (since 2006 !) digital processor was out of polarity, at least in the earliest version (which I may well have had several of at the time).

I had never tested it before, I just couldn't believe it would be wrong, especially through the digital connections.

Even this time, it took a long exploration of other ideas before I finally saw with my own eyes that it was a simple polarity difference, not so much a difference in latency.

So while I named 'Eisenson's Revenge' the kind of issue where you are at some moment having made some major mistake, but thinking about so many other things you haven't caught it yet, not so much that you are deaf or whatever, this isn't really that.  In this case I presumed my device to be correct for 18 years.  I even used it to do polarity experiments!  (However I failed to even hear a difference, so it wouldn't matter if I had the conditions reversed.)

George was constantly telling me I needed to test these things, everything.  Of course, test it his way, by listening, which is highly unreliable at best, if not completely random.  But he was right I should have tested this before now.  So this error needs to be named after him, when you just assume the polarity is correct and it isn't.

And further, by mixing DEQ's of different vintages over time (as units burned out and were replaced) further compounds the problem because some units would be out of relative phase with the other units, causing issues in systems with multiple ways.

Leading to the situation in the past two months, where I was adjusting with one way out of phase, so compensating for errors caused by the phase inversion, a complete waste of time.  Though it's only the points closest to the crossover frequency which seem most affected.

However, then the same is true about the sub inversion.  Which I'd have never discovered if the chairside DEQ weren't out of polarity.  In addition to the faulty DEQ, somehow the subs seem to be inverted, and have been for some time.

*****

Getting the microphone hooked up to scope hasn't worked yet.  I can't find the 9V AC (!) adapter for my M-Audio microphone preamp.  It's not clear my Focusrite is working right without a computer connected but just a USB power supply.  I've seen no microphone output on my scope yet.

Meanwhile, I have a few Oscilloscope apps on my phone, so I tried one, SignalScope X.

When I ran my "SoftPolarityTestRight" through the right channel (I don't run the LoudPolarityTest through speakers because it's dangerously loud but works ok for electronics) I was able to get an image which shows a very nice 1ms impulse looking almost square and perfect, trailed by bass and reflections

Right Channel playing Soft Polarity Test

But when I ran the same test signal through the right sub by itself...it's simply inaudible.  The test signal simply doesn't have any bass at all, it's all 1kHz and above.  The test signal does have DC since it only goes in one direction, but that DC gets eliminated in many ways, including the servo loop on my Aragon 8008 BB amplifier, and the bass transformer in the Acoustat.  For that matter, I don't think the Emotiva Stealth DAC passes DC either, and it's certainly filtered out by the crossover for the panels and the DEQ.

So I ran a different test signal, the one that goes with the SpeakerPop app.  It's not a series of spaced pulses, it's a series of very long duration (OTOH one second) sawtooth waves.  Running that, the subs produce a very weak sounding "buzz" sound.   Once again, there is very little bass at normal frequencies  20-200 Hz in such a signal.  And again, the near DC part is filtered out in many ways.  (In fact, to avoid stressing my amps and speakers which I did a few years ago with this signal, I created a "normalized" version with no DC offset and reduced amplitude, and that is what I actually played.)

Still, there was sufficient output to capture it unambiguously in SignalScope X.  But the result is not easy to understand.

Right Sub playing normalized Speaker Pop triangle wave

Is that in normal polarity or inverted polarity?  There's no easy answer.

At first I thought this showed the phase shift above and beyond the crossover frequency all the way up to 20khz, some of which might be leaking through the crossover and sub speakers.  But that doesn't make sense because such output should be too weak to measure, I'm barely able to measure anything at all.

But then later I realized this is showing the effect of room reflections on the bass.  Bass tones are so long that you can't even play one full cycle without the room reflections butting in.

I verified that the source of the wrapped sub tones is NOT the crossover or DEQ.  I viewed the analog output of the front sub DEQ (which follows the crossover so that's in there too) on the scope, it it was a low frequency sawtooth, just one positive impulse fading into the background on each cycle.

Speaker Pop electronic output from Sub DEQ

Studying the sub output again, I pondered whether I should consider the very first stroke as the polarity of the output, the first large stroke, the last large stroke, and so on.  I couldn't make up my mind about this.  But then I also noticed that the relative widths and sizes of the up and down strokes varied a lot with position in the room.  If you were taking the first large stroke, or something like that, the polarity would invert going from very close to the subwoofer, to the listening position, and then once again to the kitchen doorway in the back of the room.

Right sub Speaker Pop at different location

Right sub Speaker Pop at different location

Which is making all these stroke things seem like reflections.

Well, how can you tell what the polarity is with a jumble like that?  I should note that I got similar results with the Speaker Pop app itself (which uses this same test signal).  I got one polarity near the speaker, a different polarity at the listening position, etc.  So maybe the app is having the same trouble here as I am in interpreting the waveform because of all the included reflections.

Way back before subwoofers, I first learned to test the polarity of speakers by connecting a battery to the speaker terminals.  I think my friend Bro showed me that.  I was leery of the idea, thinking the battery might harm the speaker since the speaker had very low impedance at DC.  Bro assured me it was not a problem, the battery itself would be limited by its own internal resistance to not put out enough current to damage a speaker driver.

But I can't do that with subs now, it seems.  I can't just connect to "the speaker terminals."  I must go through the very fancy electronics that forms the "plate amplifier" of the subwoofer.  The latest generation on my right sub has only line inputs, balanced and unbalanced.  If there were speaker terminals, they would just feed in through the amplifier in the same way, but with more attenuation.  And most certainly the amplifier does not pass DC, etc.

Now the other way of figuring out the correct polarity was to see, experimentally, which gave you the loudest output at the crossover frequency.  I worry that might not be correct in some cases, but generally speaking it should be correct.

*****

I tried using the Speaker Pop app which was easy because I was already playing the (normalized) Speaker Pop tone 

The Right channel was very inconsistent, though it mostly seemed like negative polarity close to the subwoofer, positive polarity at the listening position, and negative polarity in the kitchen doorway.

The Left channel was a bit more consistent, almost always positive polarity near the subwoofer.

I tried playing both sub channels together and inverting the polarity of the right sub.  At first, inverting the polarity of the right sub made it sound "weird" and I thought that couldn't be right.  But when I measured the "weird" sound, it was about 2dB louder.  It was more like a thrack than a tap, and I was beginning to think this sawtooth like impulse *should* sound like a thrack.

I also noticed that indeed it seemed more centered with the Right channel set to negative polarity in the Tact.

I went to the SVS app for the right sub to change the polarity there...and as I was changing it, I could have been wrong, but just before I "changed" it I noticed that it seemed like it was already in the "inverted" polarity so nothing changed when I "changed" it.  I repeated the previous tests in both the Normal and Inverted to be sure, and confirmed that the previous measurements corresponded to the "inverted" mode in the SVS sub app.  So it appears that until now I had been running the Right sub in the "Inverted" mode of the sub electronics without knowing it.

This might have happened when I replaced the plate amplifier a couple years back.  There was a wire harness and the best thing was to test it both ways.  Back then I used the Speaker Pop app and getting the usual ambiguous result, I decided that "Inverted" was correct because that was how it measured at the listening position.  Well, now I see that having the right sub set to Inverted in the SVS app was exactly the wrong thing to do, and was at the bottom of the DEQ inconsistencies and the fact that the inverted DEQ actually worked better on the right channel than the correct polarity DEQ I normally use.  But not so much on the left.  (So now I believe that the Right sub had been out of polarity, but not the Left.  The Left channel was different but not worse when I used the chairside DEQ.  When I started this post and investigation I had believed that both subs might be out of polarity.)

All these years though, I must point out, that it actually sounded fine, even with the imperfect EQ and out of polarity sub.  Making one of these mistakes isn't like leaving the water running.  It's more like it's hardly noticeable in the overall scheme of things.  And this is more true for me than usual because I'm using such a steep crossover, the effect of panel and sub being in the incorrect polarity has relatively small effect, localized around the crossover frequency, where I could easily equalize it out.

But now, with both SVS subs in their "Normal" polarity as they should have been, and using a non-inverting DEQ, I believe the polarity issues have been corrected.  (Even though I still don't have a good test signal that makes this easy to see on an oscilloscope.  There's just a preponderance of evidence that points this way.)

Speaker Pop has been a very useful tool in these regards despite it's ambiguity in some cases.

*****

It occurs to me now that I *can* use the polarity inverting DEQ as a chairside DEQ so long as I invert the phase of the particular way it is operating.  So if I use the chairside DEQ on the panels, I simply invert the panels, or I could invert the subs and the Tact to achieve the same effect (ignoring absolute polarity, which I think I can do mostly).  It's an inconvenience, and it also means I can't do fast A-B testing where I had one adjustment in the chairside and a different adjustment in the front.  But it's still possible for month long adjustment sessions, as I am just now (hopefully and finally) completing soon.

*****

Oops!  I thought it was all done, I was now going to "finalize" the PEQ's for both channels not that I figured out the polarity issue which was dooming the right channel.

So for the "End of Eq'ing" post I was going to lead with the measured spectra with the front panel DEQ's for both subs and panels.  But something was clearly wrong with the right channel again!

Right channel, no panel EQ

I hadn't re-loaded the '2024' preset into the panel DEQ.  For doing some test, I had loaded the 'Inital Data' preset, and just left it that way.

So I re-loaded the 2024 preset, and the Right channel was looking pretty good, and you can see the benefits (and/or losses) from my EQ'ing, which I think is overall a plus.  Notably the most depressed point in the upper bass, 160 Hz, is right on the line with 1kHz.

Right Channel, latest 2024 presets

And I measured the left channel, it was as looking pretty good.

Left Channel, latest presets

But then I noticed something quite wrong.  Though I had changed the polarity of the right sub at the panel, the preset still had the 14ms delay I had found to work the best before I had done that.

Re-measuring the Left channel with that 10ms (default design) delay and it looked horrible, with a huge depression around the crossover frequency.

Left Channel, Design 10ms Delay

And that gaping depression could be solved with either 14ms delay (which I had just removed) or polarity inversion, just as the right channel.

But what about my that/thrack test which made me believe only the right sub must be inverted and the other one not?

Well, instead of repeating that possibly erroneous test, I played the Phasing Test from Stereophile CD2 with the panels disconnected.

That also was very ambiguous.   It did possibly seem that with both subs inverted, the out-of-phase was more out-of-phase, and the in-phase more nicely centered.  But when I measured loudness, when I played the out-of-phase with only one sub inverted (which now seems wrong) it measured about 1-2dB softer on peak, and the in-phase about 1-2dB softer on peak, exactly as it should have, compared to having both subs inverted.

I am thinking we can't go by the phasing test here because Atkinson's bass has lots of weird harmonics and phase shifting going on.  If I switch the phase during one bass tone, the results are nearly unpredictable.  I also think some out-of-phase cases are louder than they should be because they excite certain twisted tangential and oblique room modes more.  I should mention in all these phase loudness tests, there was a significant amount of room rattling going on.

I'm thinking the numbers here which are unequivocal are that the right sub must also be inverted from it's former condition to avoid a depression in the crossover region which happens otherwise.  And the effect on what little imaging I can hear from the stereo subs is (because it's highly ambiguous) is either acceptable or slightly superior.  I suspect on low continuous bass tones around the crossover frequency the improvement would also be as noticeable as with the pink noise.

When I re-measured the Right Channel with the spurious extra delay removed (so back to the 10ms design delay which every other speaker way uses as a baseline)  it looked even better, what I thought I had achieved with the chairside DEQ before discovering all the polarity issues:

Right Channel, 2024 preset, 10ms design delay

That passes with flying colors!

In the older generation SVS left sub plate amplifier (prior to bluetooth app) there is no "polarity" control, only a "phase" control.  I'd rather not use such things especially not knowing how they work.  Is 180 degrees the same as inverting polarity?  It might be adding some internal delay instead, or performing some other kind of complex operation.  Anyway, I set it to 180 degrees, with the delay back to 10ms, and it wasn't looking as nice as with the 14ms delay.  In the right channel, no delay was as good as the real polarity change.

Left Channel, Design Delay, SVS Phase 180

I recall using the SVS "Phase" control before and determining that it was useless.  This is only barely as good as the 14ms delay version shown above and only slightly better than the 10ms design delay with no inversion.  Similar results could have been achieved by adding about 2ms more delay, which is perhaps what it actually does.

What needs to be done is to invert the polarity through the balanced cabling.  I think I had and WAS using (and perhaps still?) a polarity inverter in one of the channels.  It was mounted midway between two balanced cables so not as to hang on any xlr connection.  It will take some work to see if there is such a beast already connected now (and remove it) or needs to be added (in which case, I will have to find the old one or buy/make a new one).

Oh boy.

*****

Fortunately I easily found my 2 XLR polarity inverter adapters.  None appear to be used on the existing left subwoofer connection; though it is marked with an "S" label, I think that means "Subwoofer" and not "Switched."  So it will actually be quite easy to add the polarity inverter tonight right behind the sub DAC where I can easily switch it out when needed.

So the plan is to switch that in, change the "Phase" control back to 0, and re-test.  Hopefully then this phase (and post) will be completed.

*****

As expected, using the polarity inverting adapter is much better than using the "Phase" setting.  I'm getting near perfect bass, main problem now is in midrange depression where there's no filter (looks like it needs a boost).

Left channel, XLR polarity inverting adapter

So that's that, supposedly inverted polarity seems better in the left sub and perhaps right.  In opposition to the SVS setting (fwiw, since I did amp replacements and may have erred) and the notion that I have a "Phase Linear" and also "Linkwitz-Riley" crossover.  Perhaps it's due to phase shift on both sides or one in particular.  Perhaps the Speaker Pop app really isn't worth a damn on subs for determining polarity in the first place...and I'm not sure anything is.  And since that was what I used to set up the subs themselves, their nomenclature might now be wrong too.

(Note: I "changed" the polarity on the Right sub, but I changed it to Normal instead of Inverted.  So, nominally the right sub has correct polarity...if I installed the plate amplifier correctly...which is highly uncertain...but perhaps that should be the default assumption here...I've long used phase reversal on the left sub because I think I installed the first replacement amp wrong, but then tried to get the others right?  So, anyway, whether one or both subs are in "inverted" polarity now is a big enough question mark so as not to labor under the belief I am using inverted sub polarity...I simply do not know, but some indications suggest that, like the Speaker Pop app, and my own attempts to measure it have so far been futile and I grasp that a suitable measurement might not be easy.)

This phasing between panels and subs is clearly right this way, however.  The center is more solid, bass is more solid, etc, and yet without being "too loud" either, remaining effortless like never before because less cancellation.

Given the uncertainties here, I'm not sure how much to be disappointed that my "linear phase" crossover may not actually be "linear phase" at all.  If you have to invert the polarity of one side (and it's not actually clear I'm doing that) you cannot possibly have truly linear phase.  There must be at least 180 degrees of phase shift somehow.

If that's all there is, it's hardly worth worrying about.  That's how much phase shift you'd get from the simplest possible crossover, 6dB/octave, which also requires an inverted side.

Linkwitz investigation suggested that phase shift was not audible until you got beyond 4th order crossovers, which would be giving you 720 degrees of phase shift.

But it does raise questions.  If my crossover isn't actually linear phase, just how much different from linear phase is it?  A standard minimum phase 8th order crossover would have 1440 degrees of phase shift.  I still believe I must be nowhere near that with my "linear phase" version, but I can't prove it, and maybe it's worth more exploration.

One tack would be to develop tools to see what the polarity of the subs actually is.

Another tack would be to try different crossovers.  I could try rolling back to a standard minimum phase 4th order Linkwitz-Riley crossover.  It probably wouldn't be very different, and in theory would give me both in-phase and out-of-phase options.  Since, in theory, the drivers are in phase quadrature either connection supposedly gives the same frequency response, but the in-phase connection actually has more group delay than the out-of-phase connection, which is pretty mind boggling.  Even then, spurious phase shift from other sources including the drivers themselves might well mean, if examined as carefully as I have done here, that the out-of-phase connection actually has better frequency response too.

And a third tack would be to explore the phase shift itself.  To do that, I'd need to use much more sophisticated tools, like the Arta software I have.

I'm willing to put that all on the back burner now.  I have pretty good (extremely good in my opinion) sound already.  Improving the phase linearity of my system, which is now apparently not perfect, is not really a high priority except for my ego.  I believed I had solved phase problems, and in fact I have not.  Even if I did, without making other things worse, it probably wouldn't be a big deal.  Frequency response is #1.

On that score, there still are a few more wrinkles that call out for fixing, perhaps the biggest being the depression around 400 Hz in the left channel.

And macro adjustments such as raising or lowering the bass plateau aren't necessarily completed either.  Now that I've raised the crossover range output by "inverting" the subs, it's possible the overall bass level can be lowered again.  Only extended listening can guide that.

So while this audio project is not entirely completed, it will never be, and I think it's worth putting a marker her with all the remaining issues and ambiguities by giving this new phase alignment a name, enabling me to move on mentally at least.   I've decided to call it "Janus One."  I haven't used a crossover requiring me to make even a nominal phase inversion in over 16 years, and I didn't intend to do that when I adopted the present crossover.  But I never truly explored all those earlier crossovers in the past they way I have this time.  Perhaps they would have worked better with a phase inversion too.  I simply never bothered to test it until now, which is one more reason this is better than ever before.  I've never had measured frequency response as smooth and well configured as at present.