Rethinking Oppo noise measurement

Arguing over at AVGuide, I've realized I guestimated the Oppo BDP-95 noise even more incorrectly than previously thought.

My actual measurements were 39dB for player and 37dB for background.  One guy at AVGuide suggests that means the Oppo produces 2dB noise.  That's wrong.

But I was thinking something wrong also.  I was thinking that the noise was dominated by the louder source.

It turns out the loudness is dominated by the larger source only when the larger source is WAY louder.  20dB difference is often used as a criterion for accurate measurement.

If the increase is exactly 3dB, it means the two randomly correlated sources are equal in loudness.  (If they were correlated, it would be 6dB.)

So if the increase is 2dB, it means the background noise is actually louder than the Oppo, but not by much.  I don't know the exact formula, but I guess the extra dB accounts for 2dB of level decrease compared to background.  So the guestimated number for the Oppo chassis loudness playing DVD-Audio disc (measured within 6 inches btw) is around 35dB.

Of course, the randomly correlated and continuous assumptions wrt the background noise can't be trusted so the best way to do measurements is have the background at least 20dB lower.  But at least my guestimate is now somewhat better.

Denon 5900 manual and HDMI system

I've been thinking about getting manual for my two Denon 5900 players, especially the one intended for bedroom, now offline because it's noisy.  Perhaps I can clean and/or lubricate the mechanism to make it quieter.   Earlier this year I had both Denon 5900 and 2900 players hooked up in bedroom.  I used the 5900 player for HDCD discs (it didn't sound noisy on slow speed CD format discs) and used the 2900 for high speed SACD and DVD-Audio discs.  But my rack didn't have space for all that and the PS Audio Power Plant Premier, so I took the 5900 out to make room for it, leaving my bedroom system incapable of playing HDCD's except in CD compatible mode.  I had originally purchased the second 5900 to replace, not augment, the 2900.

Looking back through my records, I see I ordered the famously hard-to-get service manual for my Pioneer F-26 tuner from StereoManuals.com.  However their collection is not indexed for Denon service manuals, and you must send them an email.

I found the manual already listed at servicemanuals.net and ordered a paper copy from them.

Meanwhile I also ordered a new kind of HDMI transmitter/receiver set for the living room.  Although I'm still planning a re-test, it seems that one of my two OWLink systems is no longer working.  OWLink systems are hard to find, and the huge discount deals that I got in 2010 are no longer available.  Other optical HDMI systems for 60 feet are $399 and up.  The more common way of sending HDMI digital video more than 30 feet is to use CAT5 or CAT6 with a suitable transmitter/receiver pair.  There are many such units available, from cheap $30 to over $1000.  Amazon has a generic brand for about $90.  I decided to get the ETS AV961G from Markertek.  Markertek is a top video supply house, so if they carry something, that's an important endorsement.  I checked out ETS and they are a US company that has been making balun-like AV products for extending signal reach since the 1960's.  Unlike the product at Amazon, the ETS requires the less expensive and more widely available UTP (unshielded) CAT cables; in fact the manufacturer doesn't suggest using STP cables that seem to be required for the Amazon product.  Plus, the ETS unit is supposed to have an equalizer adjustment for different cable lengths, according to Markertek.  Another seller says the equalization is automatic and there is no external adjustment.

Hooking up the Power Plant Premier

A few months ago, Music Direct was selling off their B stock of the PS Audio Power Plant Premier at $999, half the original price.  In the new model lineup, you had to get the $3999 model to have the same rated current capacity.  I checked out reviews for awhile, and decided I had to have this for my master bedroom, which is at the end of long circuit that includes my computer room, which no doubt generates a lot of electrical noise.

One of my key thoughts about the Premier is that it would enable me to run my bedroom power amplifier on regulated AC power, which might impart some Krell-like qualities.  My currently non-working Krell FPB 300 has regulated power for the output transistors, a very uncommon feature.  Most power amplifiers have simple capacitor filtered power supplies for the high voltage rails, which could let a lot of the AC line noise interfere.

Unfortunately, when I got everything hooked up and turned on the Premier last Sunday, something was making a horrible buzzing sound.  I disconnected all the things on the switched outlets, and re-hooked one by one.  Only one thing seemed problematical, and that was the Yamaha TX-85 tuner I haven't used in years, but since it sits at the bottom of a bunch of highly interconnected equipment, I simply haven't removed it yet.  So it was easy to leave that unplugged.

But then I plugged in the Parasound HCA-1000A amp I had been using, and it didn't work at all.  Something about the episode with the horrible buzz had somehow broken my power amp.  I'm hoping it's just a blown mains fuse.

Anyway, this was the perfect opportunity to plug in my newer and better amp, the Parasound HCA-1500A, which had been used for awhile in the living room until I got the Aragon.  But as soon as I turned it on, the horrible buzzing came back.  It wasn't connected to anything but AC power from the premier.

Not wanted to break another amplifier, I simply decided to leave the amplifier hooked up to the Monster Power 2000 strip that powers everything not connected to the Premier.  (Both the Monster and the Premier plug straight into the wall.)

So while only intended to make one change, I had to make two: putting line level equipment on regulated AC power, and putting online a better power amplifier.  That helps to make up for the fact I couldn't make the change I wanted to.

It does seem that with these two changes, the sound now has much better depth, a layered depth a bit like the living room system.  The old 1000A amplifier was getting a touch of ripple from deteriorating power supply caps and was making a tiny bit of buzz.  I had previously determined that was not a ground loop problem, it was present even if the 1000A wasn't connected to anything but power and speakers.  The 1500A amp is silent, and that alone makes a big difference.

I can now use the PS Audio remote to turn on and off my system safely.  I haven't been able to use my macro-based X10 system to do that for the past few months because the PC that was handling X10 macros died apparently due to power supply failure.  With the Power Plant Premier, I set a delay on the outlet (actually, the amplifier outlet) that powers the DCX 2496 crossover, so it turns on last.  The DCX itself has a nice muting feature on the outputs, but you can still get a huge pop in the output if you turn the preceding DEQ 2496 equalizer on after the DCX has come on.

Initially I was noticing that both the input and output power were shown as having 2.0% THD.  Then, I just happened to plug in my original style Tensor lamp, and the problem went away (even with the lamp turned off).  Now it shows 2% THD on input and 0.5% on output, which is what one should see with correct operation.

Worried that the problem with power amp might indicate some problem with the output of the Premier, I got out Tek scope and measured through a 1M resistor.  The wave on both AC input and output looked virtually identical.

The heatsinks on top of the Premier do seem to get fairly warm, even with a mere 2.0 amps load.

I still wonder what happened to my 1000A amplifier, and why I couldn't get the Premier to power either of my Parasound amps.  Maybe it's better just power line level anyway.  Although I tend to think of my system as "all digital" there are actually lots of non-digital interfaces:

Denon 2900 analog output goes the #2 DEQ 2496
DEQ 2496 #2 uses analog input
DCX 2496 crossover has analog outputs (driven by barely adequate power supply)

Disc players should be mechanically quiet

I originally bought the Oppo BDP-95 player as my new central video player, to play Blu Ray and other discs in the Kitchen, from which digital AV signals are set to other rooms.  The fact that it has a cooling fan will not be a problem there, because the kitchen (my AV production center) is full of equipment having running fans, not to mention refrigerator.  I might have done just as well with the BDP-93 or some other player, but I just wanted the hottest new model.

When it arrived, I decided to hook it up to my main living room audio system just because it has a high end audio output section, including current best DAC chips and balanced output.  I was immediately hooked, and shortly decided to use the Lavry AD10 to digitize balanced analog output from SACD's and DVD-Audio discs which don't permit you to access the maximum resolution 24 bit digital audio directly.

But it has bugged me all along that the Oppo, which not as noisy as some old DVD players, is not exactly the quietest either.  Notably, the Oppo has a fan, which is audible, though just barely, from listening position.  I figured, however, that the mechanism itself was fairly quiet, perhaps quieter than the Denon 5900 I had been using previously.

I finally got around to comparing the mechanical noise levels of both players when playing one of my favorite SACD's, Santana's Supernatural.  And it turned out that the Denon 5900 makes way less noise, and the difference is very noticeable when playing discs.  The fan is actually the smaller part of it.

When playing this disc, the Oppo has a characteristic woosh sound, as you might expect something spinning at high speed.  It is actually a very pleasant sound, but it is just too loud for good hifi.  I tried using the RTA application of my iPhone to measure this noise, but that did not turn out to be a very good tool for measuring, and as I was trying to measure the noise level, the heater and refrigerator kept deciding to turn on and off, making it difficult to make the measurements.  The noise produced by refrigerator pretty much masks the noise from the player, at around 41dBA everywhere in the room.  When the refrigerator shut off, the Oppo seems to measure about 39dBA from about 6 inches.  When the Oppo was turned off, noise level dropped to 37dBA.  This sounds like a small difference, but from 6 inches the Oppo is clearly and plainly audible above background noise.  I decided to mainly rely on subjective noise level assessments, the SPL meter at hand not being particularly informative.

With the Denon hooked up and playing the same disc, the chassis noise level measured 37dBA regardless of whether player was on or off.  It was obviously much quieter.  If you listen carefully, the Denon noise is more complex and mechanical sounding, with a tonal whine and faint clicking.  But it all sounds about 12dB or so quieter than the Oppo, and that makes a huge difference.  From my listening position, it is much harderto tell whether Denon is running at all, except when it changes tracks and some motor seems to change speeds up and back down.

Actually listening to Supernatural was a revelation.  I had been listening to this disc for a couple weeks on the Oppo, so I was feeling like I've heard it all, but I hadn't.  I first noticed an interesting kind of coherency: purpose.  You could almost sense the two percussionists on the two sides of the room.  I wasn't just hearing the instruments, I had the sense that the percussion instruments were being played by different people, and I could sense the two different people, and what feelings they were trying to communicate.  In short, the music seemed to have purpose.  It wasn't just "there" it was being created by real people, to communicate.  Another description I thought of at the time: integrity.

Later I noticed more typical transparency, I was hearing very low level counterpoint I hadn't noticed before.

Overall, I felt the Denon provided a subtle but valuable improvement in transparency.  But everything I heard could be explained just by the chassis noise level, which is way higher than the electrical noise levels in the output signals from the players.

Consider that a typical peak playback level for me is around 95 dB (and that's generous).  Then an ambient noise level of 39dBA is only 56dB down, enough to obscure musical details, with the music rising out of the fog of whirring chassis noise.

As far as the electrical output, the Denon is spec'd at 120dB S/N whereas the Oppo is probably at least 6dB better than that (some claim around 130dB).  But this is all moot if the chassis noise is only 56dB down.

What's worse, both players are in front of the room, in between the speakers but just slightly further back.  That means that any chassis noise cannot be separated from the musical image by the brain.

I had also been thinking of moving the Oppo back to highly absorptive corner in back of the room to make the noise lower and get it out of the audio image.  That is what I had been doing with the Denon, but I often found it very inconvenient.  Junk also tends to accumulate in that part of the room, and quite often the player was not accessible.  Then I discovered how convenient it is to have my disc player in front of the room along with my other equipment.  It's always accessible, and much more convenient to change discs without getting on knees, etc., as well as operating controls and getting feedback on controls.

Although the Denon isn't silent either, I think it's just quiet enough to keep at the convenient front-of-the-room position, at least for now.  So then I have both convenience and top quality performance.  At some future time, when (and if) I'm ever more organized, I could put my disc player in back of the room again, and then the Oppo might well be a contender again.

The sound of the Denon was possibly slightly brighter than the Oppo.  This could also be explained as the Oppo transport noise masking high frequency details.

While it is a little old, there is nothing inferior about the audio design and performance of the Denon 5900.  Like the Oppo, it has a separate analog power supply for the audio section.  It uses Burr Brown's best DAC's from 2004, essentially identical to the ones in the Denon 5910 which was still being sold as new in 2010.  The ESS 32 bit DAC's used in the Oppo may be slightly better, but I suspect it would be rather hard to hear the difference, assuming the players were not contributing their own mechanical noise to the room.

The Denon output is not balanced, but I run the composite video output through an isolation transformer before connecting to the TV, and composite video is OK for watching the menus and pictures on DVD-Audio discs.  The isolation transformer makes it less necessary to have balanced audio.

One problem is Denon can't play the high rez audio Blu discs in the recent Pink Floyd immersion sets.

Busy December Weekend

I attended important social events on Friday, Saturday, and Sunday.  I was getting over the stress of construction issues with my back yard workshop.  And I got car washed and did a long overdue quick mowing of front lawn.

So I wasn't expecting to get anything done on Audio last weekend.  But I did, I could check off at least 4 items on my Audio to-do list.

Sometimes it just takes a little push to get somethings started.  On Friday night, after coming home from Christmas Party at work, I finally got around to comparing the chassis noise level from Oppo BDP-95 disc player and Denon 5900, both playing the Santana Supernatural SACD I've been listening to for last couple weeks.  The Denon was way quieter (see later post).  That was #1.  I had been intending to do this test for about 10 months, ever since I got the Oppo.

Having hooked up Denon again, I decided, why not try listening to it again.  That was a revelation, I've decided to use Denon as my living room player, and move Oppo to kitchen to become my central video disc player (which was what I originally intended when I bought it anyway).  That was #2.

Then I decided to do some listening in bedroom.  On Sunday I decided to hook up the PS Audio Power Plant Premier I had got a few months back.  I finally did get it hooked up, not without some typical unexpected difficulties.  That was #3.

This wasn't what I planned, but in the process of hooking up Power Plant I broke my Parasound HCA-1000A amp.  (Probably just needs fuse replacement.)  So I hooked up the HCA-1500A amp I bought for the bedroom, but which had been used (until recently) in the Living Room, and had been just sitting around since I got the Aragon amplifier for the living room.  That was #4.  I had to re-adjust bass level by ear because I couldn't re-measure the gain level of the now-broken HCA-1000A.  The bedroom system now sounds much better, and there had been a background hum from the old deteriorating amplifier which is now gone.

The result was both living room and bedroom systems have been significantly improved.  The bedroom now has a bit of the layered depth I get with living room system.

HDMI extensions

My whole house entertainment system features video from several sources (DVD Player, Dish box, Harddrive recorder) run through HDMI to displays in Kitchen (where all the distributed video sources are), bedroom, and living room.  Two long HDMI lines are run with OWLink fiber optic HDMI to bedroom and living room.

This had worked well for about two years until a month ago the HDMI link to bedroom stopped working.  After a bunch of tests, including swapping the receiver with the one in bedroom, I decided that the Kevlar-wrapped optical cable (run outside the house from kitchen to bedroom) had gone bad, probably when housecleaner stepped on it.  However, that turned out to be wrong.  Instead, it turned out that, for no apparent reason, the optical transmitter in the kitchen for the bedroom link had gone bad.  When I swapped that for the other transmitter (used for living room), bedroom worked fine.  Since the bedroom link is more important (used every day) I keep the good transmitter running on the bedroom link.

OK, so now my best evidence suggests that one of my two OWLink transmitters has gone bad (it blinks, while the receiver LED stays constant, a typical OWLink failure mode).  But even that is not certain after last week, when I tried to switch my good OWLink transmitter over to the living room link.  THAT did not work, so I had to use a temporary workaround for my party (I used my Oppo BDP-95 for the DVD, normally I only use that player for audio discs).

So now that it seems I have misdiagnosed the problem twice, perhaps my second set of OWLink components is actually still working, or needs something trivial like a new power adapter.  But I wouldn't count on it.

If it still works, the optical link is worth keeping.  It has one very beneficial extra feature: it does not introduce any kind of ground loop into the related systems.

Meanwhile, I've been looking at new options.  Getting a new OWLink setup is one of them.  But rather expensive and hard to find (possibly even the 3 or so websites that still claim to have them are sold out).  Until today, the lowest price seemed to be $599, but today I discovered I could get one at digitalconnection.com for a mere $399.  That's still awfully high, but still worth considering.

Other alternatives:

2) CAT-5,6,7 based HDMI transmitter and receiver (CAT-7 said to be best)
3) 2 30 foot copper HDMI cables and a relay
4) Other optical system

There are many varieties of (2).  Some of the (very expensive) ones use only one CAT5,6,7 cable.  Using just one CAT5,6,7 requires some subtle multiplexing.  Most of (2) use two CAT5,6,7 cables.  I've now found one as cheap as $70 available at Amazon.  I'm worried that if shielded twisted pair (STP) wire is required, the shielding could introduce a ground loop.  Official specs say that STP should be grounded on both ends through the connector (there is no ground pin on CAT5,6,7 wiring).

3) Here's a relay available at digitalconnection.com that permits two HDMI cables to be joined.

4) I've seen some other optical systems priced as low as $299 for 10m, $499 for 20m.  Typically the transmitter/receiver modules are permanently attached to the cable in the cheap sets, and detached in the more expensive sets, which can go up to $1500 (and I thought OWLink was expensive).  Detached is better, since you could replace cable w/o replacing the adapters.

Listening Angle

I measured the listening triangle of my most forward listening position (which I've decided has better bass...stronger bass...than sitting just a few inches further back).

The speakers are 61 inches apart (center-to-center) and 51 inches from the listening position.  That makes for an angle between 60 and 90 degrees, which sounds good for high quality stereo reproduction.  The exact angle seems to be:

angle(radians) = asin (30.5/51)  ;# 30.5 is 61 / 2
angle(degrees) = 360/3.14 times above

73 degrees

First, fix the rattles

I should have done this long ago.  The photo above was made after fixing the bookcase rattling by moving bookcases away from leaning on wall and removing junk on the left bookcase so it could be filled nicely with books and magazines, almost all the ones that were originally there.  (I never got around to photographing the bookcase before the change, it was almost unimaginably stuffed with random junk.)

The two big bookcases at the rear of the living room were rattling on deep bass tracks.  I'd noticed the rattling, and assumed it was coming from the walls.  But the walls usually were not doing the rattling, I discovered on Sunday.  It was the two big bookcases at the rear of the living room, or something on them.

I started clearing off some of the bricabrac at the top of the center bookcase, including my Fisher MPX-100 unit which I still haven't gotten around to testing.  Then I noticed a serious issue.  Both bookcases were actually leaning against the rear wall of the room.  That leaning meant that any wall vibration (and there is plenty of that) would shake the bookcase and all of its contents.  The bookcases are very heavy, but some of the nick nacks were light and flimsy, like a still-in-box Barbie Ferrari (I'm both a model car collector and a doll collector, and a red Ferrari is the iconic car of the iconic doll).  To mitigate bad karma, I also had a remote control Ford Focus on top of that.  BTW, I'm taking the Ferrari down for now, and it's been replaced by a removed-from-box remote control Tesla model car (barely visible on the top at the left edge of left bookcase in front of a white doll).

The corner bookcase would be very tough to clear out because it's jammed in partly behind my keyboard table.  But I managed to get it off the wall mainly by shifting the magazines in the bottom two rows toward the front.  That got it about 5mm away from the wall at the very tip.  Fortunately I think that's enough to handle even the worst vibrations of wall and bookcase.  Or at least it shouldn't happen often.

The left bookcase, which was not just packed but stuffed with books, VHS, cassettes, junk, magazines, etc., was unloaded down to the bottom two rows, then shifted out from the wall by an inch.  As I was doing this, I was rocking the bookcase and trying to determine where it would be absolutely free of rocking back to the wall.  I think now I moved the bookcase out too far, it is more than an inch out further out than the other bookcase.  In fairness, however, I would move the other bookcase out another half inch if I could, and the center bookcase needs to be almost this far out to make the electrical outlet near the floor accessible.  I don't plan to change it now, but I think I moved it out about 1/2 inch more than necessary.  I think when I added back all the books, and moved the magazines out to the edge, the bookcase leaned forward more than I was planning.

Keeping the bookcases from leaning against the back wall fixed the #1 rattle problem in the room.  Rattles take away greatly from the sense of dynamic range.  Of course when you fix one rattle, other softer rattles become apparent.  Another one that needs fixing is the door of the air handler for my HVAC system, which rattles on certain loud bass notes.

Now it has been written (by an acoustical treatment maker) that bookcases do not make for very good sound absorption or diffraction, which most rooms need a lot of. That may be true (and I'll think about it some more later) but when a bookcase has a rattle, it's like negative sound absorption.  Very negative to the listening experience.  Getting back to No Effect is a big step forward from that.

To be free of rattles, a bookcase should not be leaning on or touching anything but the floor.  Otherwise, when the surface it is touching moves because of acoustical vibration, and/or the bookcase itself moves because of acoustical vibration and it moves away from the surface, there may be moments where the pressure momentarily releases, and you get a rattle between the two.  Also, the surface may cause the bookcase to shake a bit, causing stuff, particularly the kind of flimsy stuff I used to have on top of my bookcase, rattle.

Now that neither bookcase is touching a wall, it is incredible how much the wall vibrates during deep modal bass notes, but the bookcase seems perfectly still.  The bookcase panels may themselves have some resonances, but they would be at higher frequencies that don't so much cause rattling, and the bookcase is made of particle board and heavy paper, which are very lossy and vibration damping materials.

If I have increased the sound level that would cause loud and annoying rattling by 10dB, it is much like (or perhaps even better than) increasing my system dynamic range by 10dB, or almost like reducing modal resonances by that much (which is very hard to do).

I discovered the need for this when testing 1 vs 2 plugged ports in my right subwoofer.  I decided NOT to bother with measurements, but to go ahead and make the change to 1 port because it is well known to be better (see earlier posts), but why not listen?  As I was listening to the 1-plugged-port case, which makes bass slightly louder down to about 15Hz, I noticed the back-of-room rattles.  I got the idea (though not confirmed) that I might be hearing worse rattling with 1-plugged-port than with 2-plugged-ports.  At the same time, I also thought I could hear the improved dynamic range of 1-plugged-port.  So I thought to myself, why not see if I can fix the rattle?

Now actually it seems to me that a bookcase can have some desireable bass trapping qualities (if not as good as engineered bass trap).  To get this, you first need to abolish rattles, because if it rattles it is worse than useless acoustically.  Then, the books, should be pulled out to the edge, to get them as far away from the wall as possible.  Heavy books or magazines should also fill as much of the frontal area as possible.  Behind the books, there will be trapped air, and behind that, my bookcases feature a lossy paper back, which serves a bit like a panel or membrane resonator.  The trapped air behind the books is squeezed in and out by vibrations of the panel, and the released energy is lost moving the books and bookcase panels.  This does in fact operate like a bass trap.  An engineered bass trap would likely use fiberglass in front instead of books to do the lossy absorbing, but would otherwise be similar.

Actually, the bottom two shelves of my bookcases are filled with very heavy paper Stereophile and The Absolute Sound magazines...probably better than books for a bookcase acoustical damper.

Accurate Stereo

In this great post, Sigfried Linkwitz describes accurate stereo reproduction.

With the listener in the "sweet spot" a virtual sound scene should open up in front of him.

Linkwitz details 3 cases, the traditional equilateral triangle with speakers and listening position, a wider listening angle of 90 degrees given by making the distance to speakers 0.7 times the width (W) between the speakers, and a long distance 2W which loses imaging detail but can be pleasant and is less sensitive to sitting off axis.

This is where I've been this year.  I started at my old 2W position, moved in for something close to 0.7W, and now am pretty close to 1W (though I suspect it's about 0.85W).  I've observed the same things he describes.

Online Jazz

A friend suggests I listen to smalls jazz club.

Discussion of Bag End E-Trap

Here's an interesting discussion of the Bag End E-Trap (which I'm thinking about getting) that wanders seriously off topic to address questions about sabins, etc., that I had been unable to answer earlier this year.

Asymmetric Port Tuning

At the very close new sweet spot (VCNSS) listening position, the image is very transparent and 3D, and the bass seems about right on Bass E (see earlier post about this recording).  Elsewhere in room, the bass is very boomy.  I thinking of getting an active bass trap for that.  I now think Living Room sounds better than Bedroom on Bass E.

Right now I have one subwoofer tuned with 1 port plug (15Hz) and the other tuned with 2 port plugs (10Hz).  I'm thinking of changing 10Hz to 15Hz, because that is the tuning most recommended by Ilkka from his distortion tests, and even SVS themselves no longer recommends the 10Hz tuning on the newest edition.

Adding these plugs has seriously improved my bass in Living Room.

But now I'm also wondering if there isn't an advantage of having the Subwoofers tuned differently.  One problem with stereo subs is that most bass material is actually mono, and therefore if played back through two speakers there can be comb filtering.  If the subs are tuned differently, there is a small amount of random phase between the two subs which could suppress the comb filtering effects.

So when I remove the second plug from the right sub, I should test not only that sub, but the whole system, looking particularly for evidence of comb filtering effects.

I've ordered a tiny amount of Auralex open cell foam.  Just to get started.  I was going to get a pair of Auralex 12" cubes, which are in an inexpensive $80 two cube kit, but decided to get 4" cornerfill (4x4x24) instead.  It seems like nobody actually stocks the cubes, they're drop shipped often with no explanation of what shipping will cost, but Sweetwater had the 4"cornerfill, which I'll call sticks, at a nice discount price with free shipping.

The sticks can be placed in the corner behind the left subwoofer.  The bass in that corner is so intense the wall and windows rattle,  and it would be great if that could be ameliorated somewhat.   I'll arrange the 4" sticks similar to a triangular LENRD in the corner.  A real LENRD would probably be slightly too large, and LENRD's are not available except in $300 sets of 8.

Auralex foam is auralex foam.  There is nothing special about the shaping on the surface of a LENRD, a flat surface would work just as well (or better) for bass.  What really counts is the total volume and where you can squeeze it into your room layout, and the more the better, stuffing all corners would be nice.

To Do List

Usually when I make a To Do List, it winds up that everything listed gets bypassed for the next thing that actually gets done.  But years later, many of the things got done anyway.  I don't take these things very seriously, they're just ideas.

1) Subwoofer tuning: though some more measurements might be nice, I've pretty much decided now that one port plug tuning (15Hz) is the best for the PB-13 Ultra's in my living room.  So I can just change to this and be done with it.

2) Deep bass tuning...short of full on room correction, which I have mixed feelings about, continuing to check out the bass in various ways and doing tuning with parametric EQ's, crossover adjustment, etc., can be helpful to the deep bass (below 80Hz).

3) Supertweeters.  Currently offline.  Put back online in far HF augmentation mode.  Adjust by ear and measurement.  Decide again if they are actually beneficial.  This year I have been adjusting HF balance by changing Acoustat angle...this is very effective, might have eliminated need for supertweeters (though I doubt it).

4) Bass traps.  Start experimenting with bass traps, possibly including active bass traps (e.g. Bag End).

5) FM Notch Blend Filter.  This has been stalled because I lost the drilled panel I created a few months ago.  Meanwhile, KPAC is now doing station upgrades, which may obviate the need, or require something different.  Last I checked, KPAC was operating at a fraction of normal power, and either stereo or mono sounded horrible.

6) Kenwood KT-6040: fix time constant to US standard.  When KPAC is fixed, I need to change put Pioneer F-26 back online as main tuner, and then use 6040 as whole-house tuner through Sonos.

7) Oppo vs Denon: do comparisons of Denon 5900 and Oppo BDP-95 playing various kinds of high resolution audio discs.  Simple start would be comparison of mechanical noise on DVD-Audio discs.

8) Move Oppo (or Denon) to back of room to lessen impact of mechanical noise from chassis drive and fan.  Use remote extender to enable nice control, and have some sort of connection to TV (hdmi?).

9) HDMI link: currently one OWLink transmitter seems to have died, but it might actually be a power supply problem.  This is not an audio problem, but I do need to fix some sort of HDMI link from kitchen to living room to watch video sources from kitchen including Cable.  Another test of the OWLink might help, or buying new kind of CAT-5 based balun.

10) Set up HCA-1500A amp in bedroom, along with PS Audio Power Plant Premier regenerator.

11) Make CD copies of a set of cassettes I have (using Nak and Masterlink).

12) Set up turntable in bedroom (using Linn, Panasonic, Behringer).

13) Krell: re-test, take apart, photograph, reassemble, send in for repair.

14) Koss phones: make adapter cable for Stax amp, photograph amp, get optimal cables for phones and install with Stax plug; disassemble and photograph second E90 unit, measure E90 bias in operation.

15) Get PSX800 turntable repaired, install new Dynavector 17D3 cartridge.

16) Replace old cap on left Acoustat with Solen (already done on right).

17) Send in Ivie IE30 for repair.

Bass even better up closer

For my monthly party last Sunday I moved listening chair from center of living room.  On Tuesday I moved it back, but positioned to the tape markings I made about a month ago, and NOT to the most recent super-forward listening position.

When I moved up again to that position on Wednesday, the imaging on the Bass E album was fantastic.

Now I've marked all the new positions (chair and speakers) with masking tape.  Other speaker and chair positions are still marked as well.


My ear now lines up with front edge of sidewall bookcase.  The slightly back position puts it in line with the second book.  Anywhere in between those positions is fine, though with the forward position I can lie back and still have great image, and I think the slightly wider speaker angle is better.  There is a slight tendency to miniaturize the image, I imagine a large 5 foot thick fishtank at the front of the room that the musicians are playing in.  But not so bad, actually, and the coherency is wonderful and brings the kind of 3d transparency that headphones can have.

While the bass does sound a bit boomy on this recording around the room, particularly at the Kitchen doorway, in the listening position(s) it settles down very nicely, rounded but not boomy.  Still, the need for acoustic or other bass treatments is still obvious.

At the listening position, living room system bass now sounds very similar to bedroom system bass, which has long been a reference after it was carefully tuned by manually set parametric filters.  I listened to Bass E on bedroom for awhile.  It is quite amazing what the comparatively compact SVS 1646 subwoofer can do.  That unit has always amazed me, I had never been so amazed by my pair of SVS PB13 Ultra's, until recently discovering the benefits of plugging at least one port.

A recent review of the newest PB13 Ultra shows that SVS no longer recommends plugging two ports with the 10Hz cutoff, and no longer provide the 10Hz cutoff.  They recommend all ports open, one port plugged, or all 3 ports plugged.

Here are some measurements of the PB13 with two ports plugged (10Hz tuning) at Home Theater Shack.  While the flatness of response is superb, the distortion gets very high at 110dB.  Ilkka doesn't recommend the 10Hz tuning except for music listening at low to medium levels.  He feels the added extension is barely noticeable but the added distortion (if you are playing loudly) is very noticeable.  Here are the measurements of the PB13 with 15 Hz (one port plugged) tuning.  OK, comparing the two, it's indeed easy to see the lower distortion at levels above 100dB.  Ilkka feels the added extension in the 15Hz mode is worth it compared to 20Hz mode (no ports plugged) and causes only a tiny loss in clean output above 20Hz.

The Rotary Woofer

I first read about it more than 20 years ago, the Eminent Technology TRW-17 Rotary Woofer.  Yes, I can't afford it (not planning to open theme park).  Peter Montcrief in the International Audio Review called it the only true subwoofer.

They have a downloadable audio test CD.

They argue that there is no clearly defined low frequency limit to human hearing.  Instead, human hearing simply becomes less sensitive at lower frequencies.  At 5 Hz, it takes an astounding 115dB simply to be audible.  For cone speakers, this is extremely difficult, it takes more and more surface area to reproduce such low frequencies.  Of course, it was exactly what the TRW-17 was designed to do.

(I'm not planning go get one, they're very expensive and require custom installation.)

Dialed back 26 Hz boost to 0.1db

On Wednesday morning, I was finding it somewhat difficult to crank the volume level on Bass Ecstacy by Bass Erotica (a very low-bass-heavy album).  So then it occurred to me that after I extended the deep bass to 16 Hz this weekend by adding one subwoofer port plug on the left side and two on the right and lowering the low frequency cutoff accordingly, I may well have been adding back in much of the circa 26Hz bass the boost had been intended to restore.

So I dialed the 26 Hz EQ (in the DCX 2496 crossover) back from 3.5dB, where I had set it most recently, to 0.1dB, effectively disabling it without removing it.

Then the extreme bass became more tolerable, and I was able to raise the level by a comfortable 10dB, making the lyrics and effects much more open sounding.  I was even dancing.  But I think the bass still needs more refinement.  I believe I can still listen to this very difficult album louder in the master bedroom, where I have custom equalized the bass with parametric and graphic EQ using a Behringer DEQ 2496.

Might be time to bring back the keyboard, which was put away on Saturday Night to prepare for a party on Sunday.  I was even thinking about using a real oscillator, but might be nicer to program a frequency offset slider into my keyboard sine program.

*****

Back on Friday night, when I found another bass port plug for my left subwoofer, I measured the effects of decreasing the low frequency cutoff frequency on 16Hz playback, and the effect of adding the one plug.  Adding the plug did increase output by a few dB, as did changing the cutoff to the one-plug recommended setting (which is 18 Hz).  Adding the plug had a bigger effect than turning the cutoff to the two-plug recommended setting (which is 16Hz).  I had the numbers written down, but they did not get copied to this blog.  Now I've found a second plug for the left subwoofer, and thinking about doing a whole matrix of measurements.  But I need to replace the C cells in my Genrad 1933 SPL meter, which conveninently has a "flat" setting and 1 inch microphone.  I was measuring levels between 89 and 100dB SPL at 16-20Hz.  I believe I boosted response at 16Hz from 89 to 92dB.  The bass plug not only increased 16Hz response, it flattened the hills and valleys between 16 and 25 Hz.

SVS shows the flattest response (ruler flat) to 20Hz with no port plugs.  They show 2.5dB loss at 20Hz with one plug, but improved extension to 15 Hz before cutoff.  They show 6dB loss at 20 Hz with two plugs, with little visible knee in the curve, but lots of drooping, so response isn't actually extended until you get down to 13Hz or so.  Three plugs produces 8dB loss at 20Hz, with no indication from the curve that this offers additional frequency response extension at any frequency.  According to their graph, I think I would prefer the 1 plug modification, very little difference at 20Hz but nearly flat extension to 15Hz.  The two plug seems to sacrifice too much response at 20Hz, but seemed better when I measured on the right speaker.



*****
Here is the 

House Curve discussion

Here's a discussion (or really, an introduction to the concept) of a House Curve, what I generally call "room curve" (coming from the Tact tradition of Room Correction System preamps).

http://www.hometheatershack.com/forums/rew-forum/96-house-curve-what-why-you-need-how-do.html

He proposes an interesting way of setting curve by making 100 Hz and 30 Hz sound equally loud.  (Just due to human hearing response, the 100Hz would naturally sound louder if played at equal physical loudness, and the effect is greater at lower playback levels.  So this is obviously increasing deep bass response over the flat or natural response, a point Wayne doesn't make or belabor.)

I find the "discussion" (actually, there isn't any discussion, except Wayne raises and addresses some issues in his own writing) rather lacking.  The rationale for house curve is very flimsy.  I suspect that HTS editors feel take house curve as a given need, didn't need much convincing.  Wayne is at his best shooting down other bogus ideas, such as that a house curve is needed because of the industry's "X curve".

Despite having this gnawing feeling that rationale for a room curve (other than flat) is at best circular reasoning, I count myself as a believer in having a house curve and Wayne's ideas (such as the 30Hz, 100Hz test) are sensible even if his arguments for them are weak.

I'm thinking the way to think about this is to consider that every room has a reflective signature, and a room curve is chosen to make music more intelligible given that reflective signature, they type of music it is, the type of speakers, etc.  Start with this as "hypothesis 1" and I think it's fairly obvious.

"Hypothesis 2" goes farther, making some specific claims.  Flat average response falls flat because it lumps together direct and reflected sound, which the brain is somewhat capable of perceiving separately.  You would think the direct sound should be the flattest, if you equalize the total sound, which includes proportionately more bass, you will make the direct sound component of it lighter in bass.

Aha, but we do have ways of mesuring or computing direct vs reflected sound.  And a system can be designed around the goal of flat direct sound.  Has been I'm sure.  And what is the result?  I don't know, hypothesis 2 could be wrong.

"Hypothesis 3" takes a different but similar view.  Instead of direct vs reflected analysis, our brains are assumed to have real-room-response correctors.  When we go into any room, we start correcting the sound to fit our perceived sense of how the room itself boosts that (in modal patterns in the bass).  Therefore, the recorded sound played back should have those same boosts.

The problem, however, with taking H3 seriously is that flat-played-back sound will indeed get the room boosts added to it.  That is what the natural room boost does to all sounds.  So from this perspective, a "house curve" would be adding to this.  But why should boost be added more to played back sound

The keyboard oscillator

For testing speakers, particularly subwoofers,  I find that nothing beats the conveninence and flexibility of a keyboard synthesizer.  Most Kurzweil "programs" (what others might call soft instruments) are very complex.  But it's not hard to cook up a sine wave oscillator useful for testing subs, building on the simple "Default Program" number 200.  Here is what I generally set up:

Tone: sine wave
Pressure sensitivity: 0
Control #8: volume
ADSR: sustain 100%, else 0%

The area between C0 and C1 (16 and 32 Hz) is interesting.  I think my EQ boost in this area helps restore strong response in this difficult region.  But each different note causes a different feature of the room to start rattling.  Only below 18Hz or so does it seem like keys do nothing.  And there I wonder if I haven't programmed my SVS PB13 to cut out too high.

*****

I can't reiterate enough how useful a keyboard oscillator is for system adjustment.  It just sits there calling me to plunk a few more notes, checking out some other aspect of the sound.  Changes I've made so far:

+5dB at 27Hz changed down to +3.5dB

Room mode cut: 44Hz -9dB Q: 2.2 (sounds all to the better)  This is the most important room mode to tame within the subwoofers operating region where it is most important to eliminate high Q resonances (the sub tends to stimulate room modes far more than the panel speakers...room resonances can almost be ignored on the Acoustats).  Back before I started using Tact correction in the Living Room (January 2010, I remember it well if not memorialized here) I had been using two tuned resonance cancellors at something like 38 and 45Hz.  The keyboard makes it easy to check these things out and be sure you haven't gone too far (I know 9dB is a lot but it is vaporized in the wind of the resonance).

PB13 subwoofer tuning: I've put in two plugs into left sub ports (leaving 1 of 3 ports open) which permits me to dial back the sub low frequency cutoff (lfc) two notches (I had previously dialed back the lfc from 20Hz to 18Hz anyway.)  So now I'm choosing the 16Hz cutoff and I'm doing so within recommended usage.  Putting the lfc in in a lower-than-recommended position now buys very little difference (1dB) at 16Hz: measuring (with GR 1933 meter) I get 69 or 70dB at 16 Hz, 69 with recommended filter setting, while 70dB is about the average response level 20-80Hz, though response clibs to 72.5dB at 18Hz.  As it now stands, 70dB at 16Hz is barely barely audible (the lower setting changes this to barely audible); the strong 72.5dB at 18Hz is nicely audible.

New house curve

I started running the Tact Room Correction analyzer for measurements on Friday evening and into Sunday.  I decided that my living room system is measuring so good it does not need full system correction (and I will need to get better about pasting from measurement into target curve).  However, I made several important adjustments.

Final Adjusted Response, Both Channels, Nov 20

First I moved speakers slightly back and out by about 4 inches, the most available with current positioning of Belkin PureAV power conditioner.  This seemed to bring a slight additional improvement in image coherency,   The highs are also adjusted to be slightly hotter, with listening position just barely off the Acoustat beam, and flat response in the highest frequencies.  The previous speaker position is marked by tape.

To reduce bass blooming around 100 Hz, I backed down the subwoofer crossover from 84Hz to 71Hz, and changed the slope from 24LR to 48LR.

I lowered panel crossover to 80Hz, not wanting to make it lower for speaker durability.  I tried several crossovers, but decided I liked the thinking behind 24LR the best.  Years ago measurements showed the LR24 as having the nicer looking impulse than LR48, but I wonder about that now and think the the Tact impulse itself has multiple cycles, so I would have to do actual impulse measurement with other program..

To increase bass in the range 22-30Hz, which looked notably sucked out in both channels, I added a bandpass filter to the Behringer subwoofer outputs, 5dB of boost, center frequency 27Hz, Q 2.2.  It could use more boost in left channel than right, but crossover currently has this set to stereo mode.

I increased delay for panels relative to bass slightly.  I get the best measurments of bass impulse when turning off crossover.  Then put sub in one channel, panel for same channel in opposite channel, can get picture of bass vs panel impulse.  I also did this method of using two channels to measure one channel for checking crossover frequency response.

Resulting delay ia now 0.85mS in right channel (was 0.70) and 0.75 in the other (there is an extra 0.10 mS delay for right subwoofer as compared with left, same as before).  This is consistent with having listening position closer to the front, so relative distance to subs from listening chair has increased slightly.

I also tried dropping the bass level from -7.0 to -8.0, but quickly decided I wanted more bass.  Funny in the crossover picture the sub bass dominates, but that's the way it sounds best.

Coherent Imaging and making the speakers disappear

More than two months ago I moved my supertweeters out of the living room.  They were contributing to a gridlock which made it impossible to move the Acoustats.  Since I am now (since early this year) listening from a position much closer to the Acoustats, it was seeming like they might be too far apart for the close-up listening position.  The angle between the speakers from my head was more than 60 degrees, and while I was still getting a center image, beyond the center things seems a bit vague,

Finally on Sunday evening I started moving the Acoustats laterally in toward the center.  First about 5 inches in on either side, then a few more inches which reached the maximum point I could move them inward because of my electronic equipment.

The new position also allowed me to move the listening chair even closer in and still get a stable center image, and moving closer in gave me nicer bass.  The nicer bass is because I am closer to both the Acoustats and the subwoofers, and because I'm moving away from the center of the room where all room modes have their main cancellation and there is a big bass suckout.

But there were two problems now.  The image started to get a shrunken quality, with the soundstage no longer seeming life sized.  And now, some instruments seemed to be playing right from the speakers themselves.

To fix those problems, I moved the Acoustats slightly back and slightly to the side.  Because of electronic equipment, notably the tuner and the MSB PAD-1 which converts the tuner output to digital, I couldn't move the speakers back more than about 4 inches.

But that 4 inches made a magic difference.  Now the speakers themselves were not longer clearly the source of as many instruments as before.  Instead, the position of those instruments moved forward, to the the same depth as the center of the image.  Thus the center of the image was no longer by itself, there is now a right-center and left-center, and much of the music appears to be coming from a plane about 3-10 feet back from the speakers.

So I'm glad I started these speaker moving experiments because I now think I may have some of the best imaging I've ever heard.  I plan to move some of the equipment so I can move the speakers even farther back from the listening chair, and possibly more the the side as well, for an even better, more lifesized image.

One thing very peculiar was that I need to dial in about 0.23ms of right channel delay to make the center image work.  Either that, or move physically closer to the left channel, so that it seems I am way off center.  I dial in this overall delay very conveniently using the Tact, though it can't be correct that way as it affects both subwoofers and speakers alike.  I tried muting the subwoofers, and I still needed that 0.23ms of delay.  Notably when I muted the subs, I also noticed that the notes in the bass line for Spanish Harlem began to sound equal, though at a much lower level.  That's very strange also because when I muted the subwoofers I have very little bass response below 85Hz where the subwoofer crossover is, and that is basically where all the first 3 bass fundamentals are.

The need for delay is very puzzling.  It might represent some early reflection, or some difference between the speakers, such as the fact that I replaced the 40 uF cap on the right side with a nice 630V solen film capacitor, but haven't made that same change to the left side.

***** Update

The next day, the need for a 0.23msec right channel delay disappeared.  I was listening to a Cactus Pear recording in which violin sounded slightly to the left, and piano more toward center.  I was thinking at first this was a demonstration of the wider center image (including left center and right center) I bragged about yesterday, but to be sure I tried headphones and realized violin (played by Stephanie) was supposed to be in the center.  Dialing back the delay to zero fixed the problem.  I recalled some of the songs I had listened to on Sunday night, and there too the need for right channel delay disappeared.

I have to believe this was either a temporary threshold shift or some similar problem with my own hearing.  It's true I was listening fairly loudly, I noticed I had heated up the Aragon amp to 150 degrees F.  But I noticed the need for delay before cranking up the volume.  Maybe it had to do with my cold, or a temporary earwax configuration.

The Bass Line on Spanish Harlem

One of the great mastering engineers had a suggestion for tuning the bass response of a monitoring system.

Spanish Harlem on the Rebecca Pidgeon album by Chesky.

One thing for sure it is a very appealing track I don't mind listening to over and over again.

But hearing the first 3 bass notes, I wonder "are they actually supposed to sound the same loudness"?  I've now tried all my systems and the Koss Phones.  All but my Bedroom system (which was actually tuned using Spanish Harlem...) play the notes in slightly increasing loudness, with the third being notably louder. The third note also has an especially "open string" quality.

Now I see what the notes are, they are G, B, D.  Indeed, it seems like the D is the open bass string D, and it should sound naturally louder.  Even if the notes were somehow played exactly the same level, we should hear the D louder due to increased hearing sensitivity.  Now, finally, I'm looking at the opening in Wave Editor, a nice Mac program, and I see that the three notes are clearly in increasing loudness, with the third being way louder the the previous two.  Here are the peak levels:

G1: -23.2dB (49 Hz)
B1: -17.6dB  (61.7 Hz)
D2: -14.1dB  (73.4 Hz)

Even those numbers, suggesting a 9dB increase, don't do the view in the Wave Editor window justice.  The D is way louder, also with the sustained ring that comes from being an open string.  The opening G looks pitifully small.

Could it be I've mistuned my bedroom system for this?  The bass line does sound very nice, probably the nicest, on my bedroom system, which has highly hand-tuned parametric filters intended to provide smoothly increasing room gain down to 16Hz.  (Has good response to 13Hz, by the way, with SVS 16-46 sub.)

Then again, maybe that was the idea, to get nice bass boosting room curve.  It turns out that actual flat response falls flat.  A bit of increasing low end boost, seems to sound the most natural (not to mention powerful) for some reason.

In kitchen I can play the synth bass in Garage Band and the 3 notes sound like identical loudness.

Parasound HCA-1500A quiescent

After about 2 hours of idling, the Parasound HCA-1500A is drawing 65 watts.  Heatsinks seem to be measuring max temperature around 108 degrees F, mostly mid 100's, measuring through the highly ventilated top cover, might actually be a bit higher with cover removed since cover may be confusing my IR probe.

After overnight idling, the draw is 63 watts.  Heatsinks are like before, with max around 107.

At low levels, Aragon amp is Class A

I was surprised to see power consumption hovering around the quiescent 160W (+/0 4W) playing KPAC radio at medium level today.  I had Tact volume set to 82.1 and Behringer level set to -6.1 (bass is at -7).

If power consumption doesn't increase, it's operating like Class A.

This might not be a surprise with many speakers, but the Acoustat 1+1's pull voltage and current unlike just about anything.

My Aragon 8008 BB Warmup

Time, total amplifier power consumption, emitter resistor voltage

1min, 140w, 21mV

2min, 160w, 25mV

3min, 175w, 28mV

4min, 186w, 29.5mV

6min, 192w, 30.3mV

8min, 190w, 30.1mV

10min, 187w, 29.8mV

?, 175w, 27.9mV

?, 171w, 27mV

20:30, 168w, 26.5mV

?,162,26.5mV

30, 158, 26.1mV

All measurements taken from right channel (what Klipsch calls "outside" channel).  While Klipsch information suggests inside channel is to be biased higher, I measure higher temperatures (by about 3 degrees F) on the outside channel, therefore I suspect my early production unit has the same bias (much higher than Klipsch specified) in both channels.

Beyond 30 minutes, the voltage continues to fall to about 24.5mV, the starts rising again toward 26.5mV peak.  After 10 hours of idling (with covers on) there is still a little oscillation, but it's close to 160w and therefore presumably 26mV.  The temperatures in the middle of the heat sink measure between 130 and 136 degrees F, it gets cooler toward the edges, down to 125 or so (but possibly also measurement error, given that my IR probe has some width function.

Thus, there appears to be a damped oscillation, the "mass" effect probably coming from thermal mass, and the "loss" coming from convection.

When playing music at moderate level, the heat sinks do not appear to get much beyond 136 degrees, and in fact cool down with music at moderately low level.  I haven't yet seen temperature above 137 degrees.

Now Klipsch called for 8mV for inside channel, and 12mV for outside channel, but that doesn't fit with my numbers at all.  And in fact it doesn't seem to fit with the Klipsch specification of 120w power consumption for 8008 mkII (same as BB) amplifier.  Looking at my numbers, an easy extrapolation shows that 120w would correspond to emitter voltage of 17mV.

From that number, or any of these numbers, given the assumption of equal biasing in both channels, we can figure out how much power the output transistors are dissipating at idle and therefore how much the rest of the amplifier is dissipating.

I'm going to use one of my best numbers, the 30 minute reading of 158W and 26.1mV.

26.1mv across 0.33 ohms is 79.1mA

79.1mA across 140V (+/- 70V rails) is 11.1w  (Note: I didn't measure rails. AC was 123V)

11.1w for 6 transistor pairs is 66.6w

66.6w for 2 channels is 133W

That means the rest of amplifier must have been consuming 25W, quite plausible.

Now that I can estimate the actual factor emitter voltage (17mV), I can calculate the maximum class A power for an amplifier with that level of bias current.

17mV across 0.33 ohms is 51.5 mA

51.5mA for 6 transistor pairs is 309mA

Maximum Class A average power into 8 ohms is 2RIb^2 1.53w (or 3.06w peak)

Maximum Class A peak power at any impedance is Vr(2Ib) 43.3w  (Vr is 70v)

Maximum Class A average power at any impedance is 21.6w

The impedance for maximum Class A power is V/I  113.3 ohms

That would seem to be the Class A specifications for a factory biased 8008 BB or 8008 Mk 2 operating at 120W idle.  It would have less than half of the 8 ohm Class A power as mine, but still well exceeds 1w.

On the other hand, an amp biased according the the Klipsch memo at 12mV inner channel and 8mV outer channel would do this:

12mV across 0.33 ohms is 

Now I need ESP950 to Stax Amp adapter

They are available commercially through A Pure Sound at $140, seems like ripoff.  They will also modify your Koss with new cable and Stax plug for $150, and that seems like a lot better value.

Here's a thread about where you can buy the 5 pin Stax Pro Plug.

It's said to be the same as 6 pin microphone connector, minus the center pin.  One manufacturer is WBI.  Allied used to sell them.  It may be called "6 pin XLRM"

Koss E-90 Energizer Inside

The inside of the small Koss E-90 energizer is fully stuffed with two circuit boards.  It's easy to get the above view simply by removing the back cover.  From there, however, disassembly gets tougher, you have to remove the feet (which would presumably need to be glued back on), the volume control, and the front headphone jack to get the two circuit boards out.  I didn't feel sufficiently motivated to do that.

The bottom board appears to be mainly power supply, and ends in a row of electrolytics that couldn't be much bigger and still fit.  The upper board appears to be the amplifier, and seems stuffed with small transistors, resistors, and other parts.  Notably toward the front there are a few mylar caps which might be easily replaced with polypropylene.

Everything I've seen continues to convince me that the power supply is almost certainly a switching design (down to the choke in the back similar those seen in all switching supplies), and most likely the amplifier is a kind of switching type (actually Pulse Width Modulation) too.  One interesting feature is the ribbon cable that connects bottom board to top board.  This is independent of all the signal connections, which are quite obvious.  Is this because the top board needs many different independent power supplies?  That may be part of it, but it could also be that the top board ultimately synchronizes with the bottom board through multiple phases of a high speed clock.

Now PWM amplifiers are VERY efficient, and usually quite capable in delivering full power at any frequency, attributes that appear in the E-90.  But where they fall down is that there is not as much resolution in the upper frequencies as in the lower frequencies, because as the audio frequency approaches the switching frequency, it can only be constituted from a smaller number of up and down choices.  This is a problem with all 1-bit-like systems, including the DSD system used with SACD, despite the 2.88 Mhz sampling rate of DSD, a huge amount of digital processing called "noise shaping" is used to shift the noise (i.e. low resolution) from upper audio frequencies into the supersonic.  Earlier PWM systems attempted to use frequencies as low as 500kHz, which is obviously way inadequate.

Although Infinity and Sony made PWM amplifiers in the golden 1970's, the idea became unpopular either for real or rumored reasons.  Bob Carver has always wanted to make the highest power highest value and highest efficiency amplifiers, and said he tried to make a good sounding PWM design, but gave up, and instead went with the rail voltage switching designs he dubbed Magnetic Field amplifiers with characteristic flair (i.e BS).  The rail voltage is switched to whatever the current signal level requires, allowing the rails to stay as low as possible, therefore requiring the amplifier to dissipate as little power as possible (most of the energy going into the load instead of into amplifier heatsinks), even less than an Class AB amplifier, though otherwise it is made just like a Class AB amplifier.  Actually the idea seems a bit rube goldberg, but Carver got it to work by using chokes to store energy sufficient to kick the rails up when needed, and that's where the Magnetic Field name truthfully comes from, but reading Carver's ads you would think of something entirely different.  Since the late 1990's, however, electronic technology has advanced to the point where very high speed PCM digital systems could drive a PWM amplifier much more accurately than previous analog systems could.  So PWM amplifiers are back on the scene in a big way, even made by high end manufacturers such as Tact and Rowland Research (the latter formerly known mainly for building massive high bias amplifiers).

But now that I've also mentioned the rail switching design, it is entirely possible that the E-90 is a rail switching design also; it would provide the needed efficiency and eliminate the need for significant heat sinking just as in Carver's Magnetic Field amps.  Koss is rather mum about how their E-90 actually operates, giving only the barest of specifications to a direct inquiry.

Stax, in contrast, has gone for Class A electrostatic headphone amplifiers.  Class A amplification is the most linear and needs the least corrective feedback to work decently well.  It is also the most inefficient, consumes the most power, and needs the most heat sinking.  But many think it's the way to go for audio because it is the highest quality.  Stax is the commercial leader in making electrostatic headphone amplifiers, it's a veritable giant compared with the tiny perfectionist operations that make amps like the Blue Heaven.  That being said, even Stax is fairly small as electronic companies go, and even smaller since the 2000 bankruptcy in which all other business lines were dropped but headphones, their most well known product, were continued (thank goodness).

Stax tube amplifiers may be the most famous and popular among it's top line products, and many feel give the right harmonic balance to the otherwise slightly bright sounding phones, Stax has made transistor Class A amplifiers for headphones since the 1970's, and many feel the transistor amps offer the greater ultimate resolution and clarity.  My view is that the Stax tube amps aren't worth the money.  The tubes used do not seem to me like the right choices.  From reading many reviews (never tried one myself) it seems the tube amps seem deliberately down rez'd to soften the music.  If you want tubes, get a true perfectionist tube design like the $4995 Blue Heaven I described in an earlier post. But there is little such complaint to be made about the Stax transistor amps.  They seem honest designs, if not as far out as some made by elves.

I have now bought a Stax SRM-1 MkII Pro for $325 (the low price coming from the fact that the seller can't actually test, but he is allowing me a 7 day trial and examination).  That is a Class A solid state amplifier similar in concept and execution to Stax's latest top-of-the-line amplifiers (which cost up to $2400), but made from 1982-1995.  It's said to be better than the current lower end Stax amplifiers, if not quite as good as the current top-of-the-line.  Picture below is from article by Ken Rockwell which you can read at this link.

More thoughts about Koss ESP950

I've been reading this great thread about headphone waterfall plots.

The author has measured Koss ESP950 and likes them a lot.  They have very flat and smooth frequency response compared to most, and very good lack of stored energy in the upper mids and highs (just some tiny wriggles, similar to but possibly even better than most Stax electrostatic headphones, which are typically much more expensive).

There is considerable stored energy in the lower mids and upper bass.  This is possibly attributable to the less-than-open enclosure.  The author strongly disagrees with the proposition that the Koss is bass weak, in fact he thinks it has a slighly dark sound, partly from the orientation of the stored energy, and partly from rolloff in the extreme highs.  However, there is rolloff in the very deepest bass.

He believes the ESP950 sound much better through a midrange Stax transistor amplifier, the 323 (which sells for about as much as the entire ESP950 package, with other stax amplifiers selling for far more).  He feels it opens up the sound compared with the Koss E90 amplifier, however waterfall plots show essentially no difference in the mids and highs.  There is one measured difference, a significant extension of the deepest bass.

I've studied some response curves, and decided Koss would sound a tad better with a mild boost starting around 2.5k, and possibly more up higher.  Tonight I tried 2dB and 3dB boosts at 3k, which is the lowest my Tact digital pre will do an easy treble boost from.  These boosts are surprisingly subtle, but both seem to make the sound lighter and have a more open quality, more like my old Infinity ES-1 headphones.  I'm sticking with the 2dB boost for now on the least harm principle, though 3dB might actually be better.  And guess what, bass instruments sound less muddy also.

One thing nice to know is that apparently the Koss E90 doesn't do any special equalization.  Other than the deeper bass response shown with the Stax 323, the frequency response is the same.

Elsewhere, I've been reading many many blogs devoted to making electrostatic headphone amplifiers (usually for Stax, but often for the Koss as well).  Often these amplifiers are very elaborate, and even the parts cost might exceed the price of all but the most expensive Stax amplifiers.  There is an enormous and unexpected number of websites and articles devoted to such projects.



Above is picture the first tube amplifier from about 10 years ago by Kevin Gilmore, who went on to design others.  He complained that the Stax SRM-T1S tube amplfiers use 6FQ7, which isn't really up to the job.  He uses 654A's.

Above is pictured the prototype of the famous Blue Hawaii amp also designed by Kevin Gilmore.  It's a hybrid transistor/tube design with EL34 outputs with current source loading, and power supply in separate chassis, and the whole package looking mind boggling.



Above is the commercial version of the Blue Hawaii and another electrostatic amp.  Kevin Gilmore also designed the KGSS (the KG is Kevin Gilmore again) can be ordered from HeadAmp!  Well, actually the KGSS does not appear on the order page, but the Blue Hawaii does, as does a fancier but ultimately not as serious looking all tube amp, the Aristaeus.  You don't actually order these because they are cheaper than currently available Stax units, the Blue Hawaii, for example, appears to run $4995.00, and they want 1/4 down payment up front, that's about double the price of the top Stax tube unit, but it looks worth the extra cost in sophisticated parts and design (read the Blue Hawaii link above to hear Kevin Gilmore describe it).  HeadAmp says they make their own products in the USA and can customize just about anything.  I suspect they'd be able to build a KGSS if you really wanted it.  It appears that unlike Gilmore's prototype, the commercial Blue Hawaii has two pairs of outputs for two heads.  Four EL34 tubes in all.

One much simpler change involves merely changing the AC wall wart.  The Koss E90 amplifier is powered by a small 9VDC 1amp wall wart.  This provides rather high impedance power.  Unloaded, it actually puts out 12VDC, though the constant draw of the Koss brings it down.  Some people report significant improvement using lab grade power supplies instead, with DC voltages as high as 12V.  I'm wondering if lower impedance power might help with the bass response.  Most E950 tweakers, however, think a whole new amplifier is needed.  Removing the back felt is said not to be a good idea, that's the only defense against dust getting on the diaphram, a potential cause of the infamous E950 squealing that may be the number one reason units are returned to Koss.

I got my Koss a nice Stax HPS-2 stand (I was worried about spilling a drink on the coffee table and mucking them up) and Stax CPC-1 dust cover, very nice.

Class A power from Class AB Amplifiers

I started a thread about calculating Class A power from Class AB amplifiers at DIYAudio, one of my favorite sites.  It was inspired by the fact that I calculate the Class A power from my Aragon 8008 BB as 3.58 watts per channel into 8 ohms (as described in an earlier post).

(Note: the full Class AB power is 250 watts, and about 500 watts into 4 ohms, and even more into a still lower impedance, so the amp has plenty of power available, but the question is specifically how much "Class A" power the amp can produce with all transistors operating in a linear range, therefore having the most quality)

Meanwhile, Mondial and Klipsch (the first and second manufacturers of Aragon amplifiers) claimed it had a maximum of 26 watts Class A power.

Much lively discussion ensued.  It turns out my calculation of the Class A power into 8 ohms was correct.  Furthermore, if my amp used the lower bias suggested in a memo from technical support, it would have 0.8 watts of Class A power into 8 ohms.

And it turns out that Mondial and Klipsch were correct also, at least for one channel of an Aragon 8008 BB...but misleading.  It's quite possible to get 26 watts of Class A power from a factory specified Aragon 8008 BB, but you must use a higher impedance than 8 ohms, probably more like 50 ohms.  So this is not really a relevant specification, but rather a way of crafting a technically correct but deliberately misleading specification.  Also, given the lower bias, it appears likely their "Class A" power was actually peak power, not average power.  So two tricks were used to craft a misleading specification.  And possibly two more tricks, read on.

The most reduced formula for Class A power calculation is 2RII where I is the fixed quiescent idle bias current, and R is the load resistance.   Since this I could be confused with the current-available-in-class-A, which is twice as much as the idle bias current, I've now decided a better description is 2RIb^2 ("two R Ib squared").  This is an algebraic reduction of the more transparent calculation (based on P=I^2R, Ia=2Ib, and Pa=Pp/2):

2*Ib  *  2*Ib * R / 2

This formula yields "average" power (sometimes incorrectly called RMS power, see the link above for some discussion of that), that's the reason for the final 2 divisor (from Pa=Pp/2).  Peak power would be 4RIb^2.  So obviously increasing R gives higher power and reducing R gives lower power.

But the increasing the R value runs into a limit when the full Class A current available to the load (2I where I is idle current) causes the voltage across the R to reach the maximum voltage available from the amplifier.  In the case of my amplifier, that occurs around 65 volts.  Given maximum current available in Class A, 0.946 amps, and maximum voltage, 65 volts, it is straightforward to calculate peak then average power:

Pp = 0.946 * 65 = 61.49W

(R = e/i = 65 / 0.946 = 68.71 ohms)

Pa = Pp / 2 = 30.75

OK, so my 8008 BB seems to exceed the factory "spec" of 26Wpc maximum Class A power (into specially chosen resistance) by just a tad.  But the amplifier idle bias current recommended by tech support is more than 2 times lower.  So an amplifier with that level of bias could not meet factory specification even with any specially chosen resistance.  There are two possible interpretations:

1) The factory idle bias current was actually much higher than specified in the memo (shown in earlier post 12mv inner channel and 8mv outer channel), and just short of what my amplifier has.

2) The factory "maximum class A power" specification was actually a peak power specification as well as specially chosen resistance specification.

I think I believe interpretation (2).  Assuming that is correct, what would would the bias current need to be?

Pp = 26W, therefore 2Ib = 26 / 65 or Ib = 26/130

Ib = 0.20a

Therefore emitter resistor voltage for 6 transistor pair 8008bb would be

0.33 * 0.2 / 6 = 11mV

Just a tad below the "inner channel" specification of 12mV emitter resistor voltage recommended by Aragon technical support.  But it seems the outer channel specification of 8mV would never make it, nor would the same values get there with the 4 transistor pairs in an Aragon 8008 ST:

0.33 * 0.2 / 4 = 16.5mV

So even with two tricks used to get the maximum specification, it only applied to the inner (left) channel on an Aragon 8008 BB, unless the bias was set higher than specified in the memo (which is possible, it's quite possible that tech support recommended an especially conservative bias to prevent subsequent failures for people who already had one problem).

****

My Aragon 8008 BB has 3.58 watts Class A power into 8 ohms, and 1.79 watts into 4 ohms, and therefore meets the "first watt" requirement popularized by Nelson Pass, whereas a factory specified unit would not.

My amplifier is biased more like an Aragon Palladium amplifier (just slightly more, actually).  But since those amplifiers are bridged monoblocks, and mine is not, that actually gives me another advantage in Class A power at 8 ohms and less, since each half of a bridged amplifier sees only half of the total load.

So mine is actually better than an Aragon Palladium.  It's an Aragon Platinum!

(unconfirmed) Aragon Bias Adj Instructions

Here are the Aragon bias adjustment instructions, from some unconfirmed tech support document.  See my very important (of course) comments below:

2000 & 3000               4-6 mV EACH CHANNEL

SERIES                      

8008ST & BB             INNER CHANNEL-12mV             

and MKII                   OUTER CHANNEL-  8mV                  

8008X3 & X3B    8-10mV EACH CHANNEL                   

8002                            8-10mV EACH CHANNEL                   

PALLADIUM        II            INNER CHANNEL-25mV             

& 1K                          OUTER CHANNEL-20mV               

                                    W/ TOP COVERS ON-HEATSINK TEMP-118 DEG. F.

4004                            6-8mV EACH CHANNEL                

2004                            4-6mV EACH CHANNEL    

A100,A100X3

A200,A200X3

DIA150,A125X5       6mV EACH CHANNEL                

IMPORTANT NOTE:  RUN AMPS UNDER LOAD UNTIL HEATSINKS REACH OPERATING TEMPERATURE.  REMOVE SIGNAL AND LET AMP TO IDLE AT LEAST 2 MINUTES TO STABILIZE.  RESET BIAS TO INDICATED VALUES

I realize now this is a very ambiguous document.  What exactly does "run amps under load" mean, how much power output, continuous sine or music test, and into what load or efficiency speakers?  And what is the "operating temperature" ?

This is far more important that a reader might assume.  The heatsinks in the 8008 and Palladium series amplifier have very large thermal mass.  If they are heated up sufficiently, 2 minutes is no where near long enough to cool down enough to approach the stable bias level.

An hour or more  might be required.  While I determined my outer channel emitter resistor voltage approaches 26mV in a long term idle warmup, when I run my amp for 30 minutes at moderate level into killer load (Acoustat !+1's) it gets hot enough to roll the voltage to 14mV for a many minutes; I watched it for 15 minutes or so and it hadn't significantly moved upwards.  This is because high heatsink temperature causes the bias compensation transistor to cut back the idle current to some minimum level.  Only when the heatsinks have cooled down sufficiently is the normal idling bias restored.

So if you start out with heasinks at a relatively high temperature, and allow only two minutes, what you may be setting is actually the minimum bias level, not the long term level.