The Plot Thickens

I found I could eliminate all the rattles in one speaker playing Grouse by lowering the level to -14dB (0dB is approximately amplifier rated output and speaker maximum undistorted level).

Grouse We want to be loved is a veritable audio torture test, as it turns out, with extremely high peak levels, probably loaded with inter-sample-overs too (where the output level goes above 0dB between samples after oversampling...I've noted other post-2000 albums that do that a lot...this can add up to 4dB to the actual output level, I've seen so far) too.  Even though it doesn't sound like the bass or anything is hitting the stops, on this album, it actually is.

I thought of it as an amazing, and pretty good recorded album.  But I tried listening to it at the level of -14dB where it seems no speaker anomalies occur.

At this lower level, the album goes from interesting to uniquely magical.  The transparency of some recorded bits is mind blowing.  There's a bit (perhaps overplayed) that sounds like an old tube amplifier going into high frequency oscillation.  This has an almost impossible you are there quality, but only played at -14dB.  I get the sense, momentarily, that I'l at John Iverson's house listening to his Corona plasma speaker, except that he'd be playing 20dB louder.

One also hears quite a bit of clever panning and other fakery.  It's great fun, even more fun than at -6dB where I had been naively listening the first time.  That seemed like about all I could stand, the album seems far louder than the playback level, even for an experimental rock-or-whatever album.

But thinking about the improvement in sound information at lower levels...surely that's not my perception at work.  Normally we hear more and more at higher levels.  Listening to the album at -14dB (which was very late at night, perhaps 2am, and very quiet) for the first few minutes, I was nearly kicking myself to kick it up a bit.  I argued with myself that I was goind to do this for an entire song (and then, the entire album) as an experiment, which I didn't want to shortchange myself over.

What must be going on is that the speakers themselves lose it more and more at higher levels.  Especially with torture albums which are slapping them around to the maximum extent.  There may be some kind of internal vibration, which gets more and more intense.  Finally, because of some crack or whatever in the internal supports on one speaker, causing a distinctly audible buzz.

But even before that buzzing occurs, there is already sufficient internal vibration to obscure information, leading to greatest effective transparency in the speaker-ear system at some lower potential level.

I had thought that something like this was going on before, but never had such a clear indication.  The Acoustats are like 2 speakers, one with Quad ESL like transparency at Quad ESL like levels.  And then also a head banging speaker which will play much louder, but loses that magical transparency at higher levels.

And it's why some people have gone to extreme measures with floor and ceiling braces to try to fix the Acoustat internal vibration issues.

(There is also an effect in human perception, where we lose information at higher levels.  But that would only be at way higher levels than I was getting to at -6dB.)

In some sense, the audible buzz is a warning indicator.  Turn the level down for better sound.  But I do intend to fix it anyway.

Ohoh. Rattle in the 2+2's

In the past few days, I readjusted the bass EQ using my method (sweep with oscillator and adjust parametric EQ functions to fix all the bulges).  Last weekend the right sub wasn't even working (not sure how long) but after fiddling with it (I suspect loose something related to AC power) it started working again and I'm now keeping it in the "ON" rather than "AUTO" mode which might keep it warm enough to keep from dropping out again.  The faulty speaker is near the doorway were it may get more humidity.  Before I replaced the plate amplifier about a year ago, I had always kept this unit turned on.

I did the EQ adjustments (notch filters) basically from scratch, staring with the biggest bulge (either 40Hz or 45Hz depending on side) and working on down.  I can't remember the last time.  Anyway, it worked out fabulously, better than ever before, very flat bass response and no 100 Hz bulge (which has been one of the toughest to deal with).

So, on Friday I was enjoying some of the most transparent sound I've ever heard, if you can call it that playing electronic music from William Orbit and Grouse.  But then, playing Grouse (which itself is full of "deliberate" rattles and distortion) it became clear there is a rattle in one speaker.  I reversed channels and it was still there.  I tried squeezing the speaker (which seemed to help at first), slight shaking, etc., nothing helped.  I turned out lights and did not see any sparks or smell anything.

I isolated a 15 second piece of music which curiously doesn't sound loud but has a very high peak to average ratio and seems to include a bunch of tones which harmonically beat somehow and stimulates the problem consistently.  Playing that over and over, it seems I can sometimes get the rattle to go away, but then it comes back again.  Sometimes playing at the loudest possible level actually suppresses it longer--but then it starts again.

I'm pretty sure it's "mechanical," a broken stator wire support or something like that.  It probably wouldn't be hard to fix, once I get the stock off of the speaker, which I've never done before, mostly fearing the part of putting the sock back on--which is near impossible to do as well as the factory did it (no wrinkles, etc).

I'm not going to bother fixing this until mid January at the earliest.  It's another "opportunity" to do something I've never gotten around to doing before.

Acoutstat 2+2

I haven't written this up yet?  The Acoustat 2+2's that I set up in July have of course made the biggest difference in 11 years, since I got the Acoustat 1+1's.  Speaker upgrades are big.

I had for long doubted the 2+2's were even better than 1+1's, for the reason that the 1+1's are more like a line source, or so it seems at first glance (more about that later).  So, I reasoned, the 1+1's are the perfect speaker for my small space, because I don't really need more dynamic range (not really, as it turned out) just the thinner profile which works better being closer to.

So, even the likes of Mr Acoustat, a late longtime fan on DIYAudio (and no connection to the original company) preferred the 1+1's, for his mostly smallish rooms.

However, as time went on, I was beginning to think there was a certain loudness I really could not (or was never tempted to ) exceed, because the 1+1's were sounding more closed in, so I always kept the levels somewhat below the 1+1 maximum output levels, so I became more interested in what the 2+2's might actually sound like, or even thought of getting a high output level 2nd set of speakers just to hear "louder" when I really wanted to.

It was about that time, 3 years ago or so, that a friend acquired a pair of 2+2's.  I heard them in his much larger room, and I was blown away.  I felt there was nothing lacking at all, they were simply better in every way.

It was exactly that same pair that came up for sale, with an interface refurb and mod (because the owner feared he might have damaged them).  They still seemed OK if not as glorious as I had remembered them, so I bought them.

It became clear after a day that one speaker had far less treble, and I couldn't even adjust the difference out using the level controls, nor did the "air" mod help.

Fearing the worst, I went through a process of system substitution, and narrowed the fault to one interface unit, and then to the HF transformer.

Well, as it happened, I had a spare NOS Medallion HF transformer to replace the HF transformer.  The rest of the transformers in these 2+2's are NOT Medallions, as I had hoped.  But actually, this makes LESS difference with the "C" mod that has been performed on them.  The C mod reduces HF transformer saturation, so the enhanced anti-saturation construction of the Medallion transformers is less critical.  These units ALSO got the "air" mod and switch added, but the "air" mod seems to bypass all the attenuation, possibly putting more voltage on the HF transformers, and frankly I don't like the way it sounds anyway.  The C mod design, and with the variable attenuator knob type interface, seems like the best interface design to me, and can be dialed in to perfection (though I have only moved the controls a hair off center) with ease.

I had almost finished re-connecting the new transformer, I was on the last lead when I noticed the original had not been wired correctly.  The - HF and - LF leads were reversed.  Well, that for sure was the source of the problem!  But having only one more solder connection left to finish the job, and compared with hours of work (possibly breaking something) removing the new transformer and re-installing the old transformer.  So I decided to leave the new transformer in place, and now with the correct wiring too.

This restored the amazing sound I had originally heard years before.  It was simply better in every way.  Even the "line source" issue is not as I had imagined.  And, it doesn't seem that having one Medallion HF transformer significantly unbalanced the system either.

Many things are at work here.  First, having twice the membrane area means that for any SPL level, there is half as much panel movement, therefore half as much distortion from that mechanism.  And there's less saturation in the transformer as well, and half the level from the Amplifier is required, meaning less distortion also.  So, no matter how loudly you play, having twice as much panel area gets you half as much distortion, and this seems apparent in the effortlessness of the speakers as well as the greater information.  That was the key thing I noticed from the very beginning (and in fact before I fixed the transformer: more information.

It's much like that 70's photo of the listener being blown away, as if the speakers were something like a hurricane.  But in this case, there no physical force or movement of one's body or hair, it's a change in one's mind: there's more there there somehow, at every level, it's more real and satisfying.

Greater information at all levels is the biggest improvement, but there are others.  Perhaps the second is that I have finally adjusted the highs correctly.  With the 1+1's, with their newer style box, one can't just "turn a knob" to change the HF level.  It's not that hard if you have the interfaces unscrewed, as I always did.  But it was hard enough, that I never even bothered to try.  I just left the HF adjustment alone, figuring it was "correct."  But I have no idea if the previous owner adjusted them.  It didn't look like he had, but also it seemed my resistance values were significantly lower than TheAcoustatAnswerMan at DIYAudio said was the factory default.  When he said that, I didn't believe it, because my boxes looked so untampered with.  But now, realizing how the 2+2's sound at about their central position, which appears to be right (for them) but midway in the HF attenuator, I realize I was playing the 1+1's too hot.

So this improvement is actually something I could have had with the 1+1's, I just never tried it.  BTW, the 1+1'\s require a different attenuator setting than the  2+2's, precisely because there is also a LF transformer setting change which gives the LF transformers less "gain."  So, by design, the 2+2's are set to a "central" attenuator setting, whereas the 1+1's are set much higher, according to the factor spec I have heard, just not as high as mine were set, which was about as hot as they get.

Now, I had compensated for the too-hotness of the 1+1's in a peculiar way, which negated the problem for serious listening (actually, NEGATED it too much, I always thought, but it was hard to do otherwise without creating a super tiny sweet spot, head in a vise) but meanwhile meant, for casual listening every where else, I simply got the too hot highs in even greater measure.  And it turns out a lot of the listening I do is casual listening, even from other rooms, because I like the way that works out as opposed to having background music piped in to every room--because then it's too much in your face, it's actually, as I'm quite honest with myself in admitting, a form of information reduction because quite often that's better for background listening.

The method was this: I was listening far off of the central axis of the speakers.  On the speaker axis, the sound is way too hot and even slightly brittle, and very much head in a vice.  Of axis a little, it gets a little better, but the head in a vice got better and worse.  Finally, between 18-28 degrees off axis, there's a range where the high attenuation seemed about right, but all the venetian blind effect was gone, all of it.  So it seemed the best positioning was about 23 degrees off axis, giving full relief on either side from that pesky vice.  That position was also, a bit too much attenuated at the listening position, but it was OK, especially with my super tweeters crossed in at 20kHz but contributing lower.  It was often unnerving, however, to have the highs bright (and sometimes uplifiting because of that) and then sit down and have the highs greatly collapse, even if actually being somewhat more natural but from the other side.

Well, now, I have none of those issues.  The 2+2's are just fine head on, and may be optimal at a more useful angle of 12 degrees or so off of the speaker axis.  The brightness with the level control centered is just about perfect at the listening sweet spot and everywhere else.  This is big, very big, for my casual lifestyle.

So part of this was my previous ill-adjustment, and the other part was something I'd never even imagined before.  The 2+2's are two vertical columns of electrostatic elements, and those two vertical columns are in fact themselves angled slightly from one another and from the central speaker axis.  That slight angling has many useful effects.

For one thing, when one is right on the 2+2 speaker axis, neither element is facing you directly.  As before, I believe these Acoustat electrostatic drivers seem to have a slight issue exactly head on, by themselves, but here they have no issue on the speaker axis because then you are off axis by small amounts (5-10 degrees I figure) which is right where the Acoustat panels my themselves sound best.  But then, as you move off the center of the 2+2's, you get more on-axis with one panel, and less on-axis with the other panel.  The blending effect of that eliminates the worse beamy qualities at every angle, at least in the range of plus and minus 25 degrees.  So now,  I can pick whatever angle works best for other reasons, and not be constrained by a horrible venetian blind effect right at or nearly at the center of the speaker.

Other speakers do similar things.  The top Martin Logan models have for many decades used a horizontally curved membrane.  In my opinion, that introduces many technical issues.  It was amazing that they got it to work at all, but the earliest models weren't very good sounding IMO.  Now I think the latest ones may be fine, but it's still complicated thing to do, as the mebrane must stretch and relax far more as it goes in and out.

The Sound Labs use lots of little speakers arranged in a cylinder.  That has advantages, but disadvantages also.  To get the same amount of radiating area, you need a substantially bigger speaker, and it's far more expensive to make.

Given such possibilities, the Acoustat way of simply angling the two columns a speaker is amazingly elegant and works amazingly well, IMO.  Perhaps, overall, it's the best approach.  It can't be done for more than two columns of panels, and work equally well.  Two vertical columns is the optimal number, as each can be only a few degrees off the central axis.

And that applies to the third improvement.  It could even be that the two angled columns actually produce a narrower effective "line source."  It does require a slightly greater distance from the speaker for full coherency, perhaps 4-5 feet (the 1+1's are coherent at 3-4 feet).  But at that slightly greater distance, the imaging is even better, along with more depth, etc.  The greater information also contributes to this, so the quality of the line source itself may not be much better, but it is far from being worse.  The 2+2's routinely have the spacious depth which I called layered depth that I first heard when I started using the Krell FPB 300 on the 1+1's eleven years ago.

I've continued keeping the minimum (in the back inside) distance from wall at 3 feet, actually 39 inches on the inside to the front wall and 47 inches from the outside to the front wall.  The factory spec of 3 feet in my experience is an absolute minimum you must exceed.  The distances I've listed are the greatest my living room can sustain and still be good for other purposes, as I discovered at my last party.   To make the current speaker arrangement work at all, I had to put the supertweeters on the inside, which is not the best location for them, but works well enough, and allows the 2+2's to be as far apart as possible, which is helpful in my relatively narrow room, and allows for the slightly greater distance from the speaker to the sweet spot, which is now just behind the room center (as I had discovered this year, works perfectly fine with the EQ I had been using, and even better when fully dialed in).  It took a month to find this new slightly further back sweet spot and angle the speakers just right, and then some more time to reset the midrange EQ's to notch out the slightly different LF resonances and the make the usual 2-8kHz slight softening for best sound.

And any amplifier has to work half as hard into the punishing 2-ohm minimum load, which is another big win.  Now I no longer need a 1200W into 2 ohms amp like the FPB 300 (which has now been retired, after one too many failures after professional repairs, when I have time perhaps I'll learn to fix it myself).  The Hafler 9300 sounds equally fabulous without any strain at any useable level.  The Hafler is somewhat power supply limited (compared with the Krell) but with the 2+2's it's become a non-issue.

So, it's a win-win-win-win!!!!  And now I have the incredible 1+1's to use as my Laboratory speaker (though that took clearing out all the junk, which could not have been done in months had I not already been retired).

Turntable 101

A friend asks about buying his first turntable.  I've decided to spell it out here as a permanent contribution to audio buyers.  (Can a person "live" without a turntable, and just get by with digital music.  Certainly, but I'm not going to address those questions here.  I have several turntables myself and would not give up playing vinyl for anything.)

Many people have first heard records played on an all-in-one player, possibly their own.  With very rare exceptions, such as systems made by B&O, these are not very good.  To get something really good, you generally have to buy a few separate parts, though a specialty store may be able to sell you everything at once and perform all the assembly required.  The parts required are:

1) Phono cartridge.  The best cartridges are generally either moving magnet or moving coil.  Cheaper systems use ceramic cartridges which are not very good.  There are also a few exotics like strain gauge that are barely worth mentioning except to audio fanatics (and yes, I have a few myself, in my collection of exotic stuff).

Moving-magnet cartridges are generally cheaper (though some still get very expensive) and more popular.  They generally have a warm sound that is pleasant, but sometimes seems lacking in detail.  Moving-coil cartridges are more expensive (but the least expensive ones are not that much more costly than above-average moving-magnets and still highly respected) and generally only used by the most dedicated audiophiles.  At best, they can have a neutral sound that is highly detailed.  Critics may say they have too much highs.  I myself used moving-magnet cartridges for the first 10 years of my own record playing practice, then switched to moving coils, which I like better.  One of the least expensive but still highly respected moving coil cartridges is the Denon 103 ($300 brand new).  Dynavector also makes highly respected low cost moving coils, which often have high output.  I have used Dynavector's most expensive (but lowest output) cartridges since the 1980's (now $1300, my first was a donation).  There are hundreds of other brands, and technical variations too complicated to describe in a short article.  I can't really advise on moving magnets, but Grado is a well known brand, and makes cartridges available (refurbished) for as little as $50.  Technically Grado calls their design moving iron which has some advantages over a conventional moving magnet.  Rega, Shure, Stanton, and ADC are a few of the vast number of respected brands.

As with most everything in audio, some phono cartridges, especially moving coils, get astronomically expensive ($20,000 and up).  But there are reasonable alternatives as I have suggested which work quite well down to $50, and often the ones that are not-quite-astronomically expensive are available used at much lower prices, and may still work as good as new.  Buying used may be part of your audio strategy (it has nearly always been for me, and though I made many mistakes, but I could not have scaled the audio heights any other way).  The stylus on a phono cartridge can wear out, however the best used properly will last 2000 hours of usage or more.  I have never noticed wear on any of my cartridges.  The cartridge internal damping has failed (after 30 years) but the stylus did not wear out, at least in part, because I really haven't played records THAT much in the past 30 years--though I'm playing them now more than in the 1980's.

2) Tonearm

The tonearm is the device which supports the phono cartridge above the record surface.

Most often, the tonearm is part of the turntable system, but many of the best units are equipped to mount other tonearms, and most often the best tonearms are sold separately, and mounted to the best turntables, which often don't come with tonearms.

One of my systems, a Linn Sondek LP12 Valhalla with Ittok tonearm, came as a package with the Ittok tonearm also sold by Linn (made in the 1980's by some Japanese contract manufacturer).  Theoretically, I could mount many other tonearms onto the Linn armboard, but I haven't bothered.  Instead, I have defied Linn guidelines and traditions by modifying my Ittok arm, coating it with Hockey Tape to provide internal damping.  Other famous tonearm manufacturers, like SME,  use sophisticated damping systems.  I would love to have a recent SME tonearm, which are quite expensive (up to $10k) but what I can afford to do is modify my Ittok, which is still a highly respected arm (but also, following Linn's approach, completely and absolutely undamped, which I don't think is a good idea).

I did buy a separate tonearm for one of my turntables, a Jelco, but I haven't assembled it yet, and I've used separate arms with mixed results in the past.  I would say it's the way to go, but I haven't fully gotten there myself.

Quite common on older Rega turntables, and others, are the Rega RB250 and RB300 tonearms.  They are quite good, but not as good as the best.  One of my friends has a modified RB250.

Some tonearms are linear tracking.  This is not a bad idea, but may suffer in the implementation, and they tend not to be as reliable as simpler pivoted arms.  I once loved my best linear tracker but haven't repaired it since it's second breakdown.


3) Turntable

As stated, the Turntable usually includes the Tonearm, described above.  The rest of the turntable is the spinning platter suspended by some kind of plinth which sits atop a rack, table, or shelf.

Inexpensive but still excellent turntables have been sold by Rega and Music Hall for decades, and now they have a number of competitors such as Project.  Famous old but still available brands include Linn, Denon, Pioneer, and Thorens.  Linns can get very costly when equipped with all the options, which are only recommended for Linn fanatics.

For quit awhile, there had been a bias towards Belt Drives, including the Linn and Rega and many others.  However, it now appears that Direct Drive can be made to work just as well, as can the earlier Idler Wheel turntables.  Generally speaking, 50's-70's idler wheel turntables such as consumer Garrard, Dual, and Benjamin Miracord suffered only from a slight rumble, which can be cured using much heavier bases, and then they become perfectly competitive with recent belt drive systems, and some people now prefer them.  Brand new idler wheel turntables of inestimable quality are now available for $30K and up.  Some audiophiles feel now that idler wheels, at one time cast off as junk, are the ultimate best design, but this is one of a large number of audio cults.  I myself purchased an idler wheel Lenco for future modification but haven't gotten around to setting it up with new custom base and arm.  My own feeling is there is some truth towards the reverence toward idler drive, but I don't yet have a working one myself, being satisfied with either direct drive or belt drive for many years now, and there never has been anything wrong with a top quality direct drive.  What this means is you have many choices among the fashionable and one-time unfashionable in the used market.  In the new market, belt drive is the most common at lower prices, and there are a few direct drives such as Pioneer tables for just a bit more.

Ancient idler wheel turntables can sometimes be had as cheaply as free, but may require some basic mechanical work, if you are up to that.  Garrards and Lencos are especially highly regarded.  Fully refurbished and upgraded professional Garrards may sell for astronomical prices.

Your safest used turntable is a simple belt drive, such as the cheapest Rega models.  There's almost nothing to go wrong with it, and what there is can be replaced cheaply.  I've never owned one myself, however, and I've often gotten myself into trouble with more complicated used turntables, and so often wished I had.

4) Phono Preamplifier

Every record playing system requires a Phono Preamplfier, but most often it is a circuit in some other piece of equipment, such as a Preamplifier or Integrated Amplifier or Receiver.  Quite often nowadays, however, Preamplifiers don't necessarily include the Phone Preamplifier, you must buy one separately.  Even if your Receiver already has a phono preamplifier, a separate one might be better.  Most moving coil cartridges require extra amplification which most Receivers and many Preamps won't provide, you have to check for that feature.

Separate phono preamplifiers can cost as little as $29, but those aren't very good.  Starting as little as $100, however, they can start getting quite good.  Prices go all the way up to astronomical, for which the units may be better or not.

I think the small Emotiva Phono Preamplifier XPS-1 for $209 is quite good, and relatively cheap in this category, but the full sized Emotiva XSP-1 Preamplifier (including Phono) for $1249 is even better.  In the past I've used tube preamplifiers, but except for extremely expensive hybrid models, they are too noisy for moving coil cartridges.  Though I myself have designed and built tube audio preamplifiers, I do not feel tubes have any special magic in audio anymore, but they do have a cult following, and there are people who like the way particular units sound.

The Emotiva preamps use IC op amps.  I do not believe this is a problem because they use some of the best opamps, which are now as good as anything, I believe.  But many audiophiles believe tube or discrete transistor circuits are better, and so there is a market for these units from reasonably priced used models to astronomically priced new ones.



5) Preamplifier

The Preamplifier might be a separate unit, but it is often part of another unit such as an Integrated Amplifier or Receiver.  This is the unit that most often has a volume control and selector switch.  At one time it needed to have amplification, but often that isn't needed anymore, making the name "Preamplifier" a misnomer.  I use the Emotiva XSP-1, which includes the phono preamplifier.

6) Power Amplifier

The Power Amplifier is the part of the system that provides actual power to the speakers.  It is sometimes part of a Receiver.  My first audio systems were based around a receiver, a now vintage Marantz 2270.  After using that unit in the 1970's, I moved on to "separates," a separate preamplifier and power amplifier, and I've owned over a dozen power amplifiers since then.  There is endless audiophile obsession over different power amplifiers, and yet, in most cases, very little actual difference among them, except when you may need extreme amounts of power.

Hafler was a classic brand of power amplifier of top quality for not very much money.  I've owned far more expensive amplifiers myself, and yet I now use a Hafler because it is good sounding and reliable.  Parasound, Emotiva, and ATI also make reliable yet relatively inexpensive amplifiers, and there are many other such brands.  Expensive amplifiers made by Pass Laboratories and Mark Levinson are fine and reliable, but you probably don't need them.

7) Speakers (or Headphones)

This is where it gets most interesting, because it can hardly be summarized at all.  There are many different speaker and headphone technologies, sizes, implementations, and prices, and they all have their adherents.

If you are just starting out, you are almost certain to buy box speakers which have two or more drivers in a box.  These can be very good, and different variations exist up to astronomical prices.

But there are also planar speakers which many feel are even better.  Planar speakers can be ribbon, planar magnetic, or electrostatic.  I myself switched to electrostatic speakers 10 years ago, and I feel they are the ultimate best technology.  But I could be wrong; many people feel otherwise.

Likewise, headphones can be cone, planar magnetic, or electrostatic, and I prefer the electrostatic ones, which are generally the most expensive too.

8) Assembly

Generally, you will buy a turntable with arm and cartridge already set up by a seller, especially if it is a store.  Otherwise, assembly can be a tricky and complicated business, but one with which most record loving audiophiles ultimately become quite familiar, especially if they have a separate tonearm and turntable.

Hooking the rest of the stuff up is basic audiophile work and shouldn't be that hard.

Required Total Bandwidth of Audio System

The correct answer is, at least two times the highest you can plainly hear.

Why two times?  Actually, the highest you can plainly hear is dependent on two things, the power levels involved (and we don't want to go there) and...the bandwidth of the test system.

As the bandwidth of the test system improves, so does your apparent bandwidth.

What's important is not that your bandwidth is "met," but that any further improvement in your apparent bandwidth is not achieved by turther improvements in the audio system.  Thats...as good is it needs to be.

This happens around twice the desired bandwidth, because when you have two separate sytems...the audio system and your hearing system...to achieve any particular total bandwidth over both those entities, requires each half to have twice that bandwidth.

Although my hearing goes to about 16kHz, an appropriate audio system total bandwidth is 40kHz to allow for the nominal human limit of 20kHz.

And since audio amplifiers are only one part of the system which must achieve 40khz bandwidth, it's appropriate for audio amplifiers to have at least 100khz bandwidth or maybe higher depending on rest of system.

Marantz AV8805 surround processor revealed

I've decided it's barely worth considering a surround sound processor which does not have the ability to control the target curve for each speaker.  In the Marantz line, the Audyssey App is supported by the AV7704, AV7705, and AV8805 units, and not earlier models such as the AV8802a.

My favorite audio tester, Amir at Audio Science Review, measured a Marantz AV8805 and did not like what he found in this $4500 unit:

At 4 Volts RMS at the XLR main outputs:

1) 91.2dB SINAD at 1kHz (0.0027% THD)
2) Shocker: THD rises dramatically at 5kHz and above, above 1% just above 10kHz

Also he also thought the jitter spectrum looked relatively busy compared to his favorite DAC, but not so much as to be an audible issue.

Now, in my estimation, Amir did something a bit unfair to Marantz, but often addressed by commenters.  Amir didn't bother to do any measurements at the official AV8805 Rated Output Level, which is 2.4 volts.  He said that was simply a crutch that Marantz used to get better measurements.  4V is the defacto industry standard level for XLR for consumers, and Marantz should be expected to perform well at that level like any top product.

Furthermore, in long discussion which compared the specs and measurements of other Marantz processors done by others, it appears that the 91.2dB SINAD is consistent with measurements done by others at different levels.

For example, Secrets of Home Theater found the AV8802(a) to have less distortion at 2.0V, and more distortion at 5V, which was still below the apparent maximum output.  Amir reiterated many times that he turned the level "down" to 4 volts, though he never indicated what he turned it down from.

This other review of the AV8805 didn't do a single measurement, mainly talking about features and giving subjective reactions to various discs.

After much discussion, it appears that the Marantz AV8802, AV8802A, and AV8805 have the same output circuit and cound be considered comparable.  The specs are identical, and in fact the specs are also identical for the AV7705 and AV7704.  Therefore, conveniently, a review of the audio raw performance of one of these units is equivalent to a review of another.  Therefore, despite many people complaining about Amir only doing one measurement (not true) and calling it a day (apparently true, he shipped the unit out to make room to walk around in his laboratory), we can fill in the gaps by looking at measurements done by others.

Though it's not great, I don't think the 0.0027% distortion is that big a deal either.  Sure, my Emotiva Stealth DC-1 (another product Amir does not like) measured 0.0003% THD in my testing, which is about 10x (20dB) better, and Amir's favorite dacs (I now use the cheapest one he recommended for measurement purposes) do even better, but few recordings are going to be made that well and no speakers would even come close.  Many highly regarded audio products have no better than 0.1% THD, which is often suggested as a limit to the audibility of nearly all distortion harmonics.

I think, and what unfortunately to this moment has not gotten enough discussion is the #2 Shocker.

Amir did some further investigation of this.  First, he ran a 10kHz signal and found a ginormous distortion peak approaching 1% at 34kHz.  Since this is not an even multiple of 10khz, this got him thinking it might be an aliasing artifact.  So, he runs a white noise signal and then discovers only 3dB rolloff at 22.05kHz, where (according to Shannon et all) there should be infinite attenuation from a reconstruction filter.  Acceptible levels of attenuation were only ultimately achieved near 40kHz.

Amir looked in vain for any discussion of digital filtering in the manual, or ways to select different digital filtering options.  Failing to find such, he pronounced the AV8805 flat out broken.

Such slow reconstruction filters are not at all unusual on audiophile gear nowadays.  There's quite a strong following for no filters at all (NOS).

Now the only piece that leaves me wondering about this, even after reading every post in a 24 page discussion (which was quite interesting in itself) is whether the high levels of HF distortion would essentially disappear if Amir had stuck to the Rated Output level of 2.4 volts.

Amir simply believes that such an output level is not worth bothering with, 4 volts is the defacto standard, he tests everything balanced at 4V, and if a unit can't do 4V well that's simply another way that it's broken.  Commenters suggest that 2V is more than adequate for many amplifiers and challenged him to suggest one amplifier where this would not be true.

Here's one aspect which wasn't discussed.  We can't simply tell the maximum input voltage an amplifier can use from its rated input, sensitivity, and/or rated power.  Most power amplifiers have considerable headroom.  So even if it only takes 1.5V input for such-and-such an amplifier to reach rated power, it may be able to handle input peaks up to nearly 3V without clipping.  This is a good reason for preamplifiers AND surround processors to have at least 4V output, as Amir suggests.

Amir does note that the level of the 10kHz distortion products including aliases does go down at lower levels.  And therefore, he admits, the audibility of this problem probably isn't that huge either.  But it represents unacceptible performance for a flagship product.  I agree.

But I'd also just like to know, if the 2.4v output level fixes things.  It seems OK in the AFAIK identical audio circuitry of the AV8802a, as tested by Secrets.  They measured 0.002% THD at 10kHz with 2 volt output with 44.1kHz sampling, and lower levels with higher sampling rates.  That suggests the Marantz would easily meet it's specified 0.008% THD from 20-20khz at the rated output level which Amir refused to test.

Digging a bit deeper still, Secrets also tested 10khz up to 5V with 24/96 data, with distortion at 0.0036%.  Unlike Amir, Secrets did lots of different tests with 10kHz and with pairs of frequencies.  Curiously, however, they did NOT report a test for 10kHz with 5V at 44.1kHz, which might speak to the high levels of distortion Amir saw at 4V output with 44.1kHz signal and attributed to aliasing.  At the higher sampling rates, distortion reduces at 2V also.

Looking at these high frequency and IM tests, the profusion of products does indeed look like it may fit an aliasing theory.  It still bothers me how it seems fine at 2 volts (and possible 2.4 volts) but then distortion products including aliasing products just take off.  It seems to me that aliasing products should be there all along and increase linearly.

It was a similar output level limitation that led to my finally ridding myself of my two Integra Research RDC-7's today.  When I bought these a few years ago I was hoping they'd make excellent DACs, which I could even use in my laboratory.  My first naive measurements at 4V showed such massive distortion I figured my first RDC-7 was broken.  It was only later while testing my second unit I discovered that distortion looked basically OK (just OK, perhaps about 0.005%) until 2V balanced or 1V unbalanced.  Then, distortion products began rising linearly from the noise floor.  At first I thought it must be clipping.  But clipping distortion products would rise much more quickly.  It almost seem like some kind of soft clipping was being applied.

All these high end surround processors are THX licensed.  This means they pass proprietary tests by THX.  We don't know exactly what these tests are, and what they required, because they are not public information.

I wonder if THX virtually demands a kind of soft clipping be introduced in the pre/pro above 1v unbalanced and 2v balanced, in order that amplifiers or speakers themselves not be the clipping elements.

Anyway, the way the shocking issue appears in all the information presented does not suggest that soft clipping alone is the source.  The products that appear in high frequency and IM tests look a lot like aliases in some cases.  However, the rapid rise when levels exceed 2.4v suggests that aliases alone are not responsible.  Then it appears the issue may be the combination of two things, the very slow reconstruction filter AND a form of soft clipping that starts just above 2.4V.

Knowing all this, I've somewhat lost interest in the AV8805, which no longer looks like a reference product and isn't worth the high price. (If you want reference, apparently the Trinnov is the unit to get, at 4 times higher price.)  However, the AV7705 may be OK for the features, and knowing that the limitations aren't likely ever to intrude on the auditory experience.

Amir correctly blames the lack of competition in the pre/pro world to the introduction of HDMI.  Apparently only 2 companies make the require HDMI chips, and they aren't interested in supporting small time manufacturers.

I've never liked the idea of squeezing my audio signal through HDMI, and now it appears (increasing distortion, jitter, etc) that avoiding this would have been good.  Given the current market, however, it's unavoidable.

Much of this impasse has also been caused by the myth of perfection.  Everyone wants "perfect" digital inputs and outputs to things, as these are considered "lossless."  One thing that's lost but barely remembered is openness.  Analog inputs are fully open, transparently, and not subject to being locked down by greedy bastards.

If instead of demanding digital I/O, we had stuck with analog I/O and just made it better, we would not be trapped by greedy chipmakers and other proprietary schemes.

I strongly believe that digital conversion is virtually transparent, especially at 24/96, so there would be no harm in sending analog audio signals from CD player to preamp, which could convert them to digital for processing, and back to analog for amplification.

This is the kind of thing I have done for over a decade now for stereo.  In some cases, curiously, I find the resampled analog to be BETTER than digital.  In not one single case have I found it to be noticeably worse.  Sometimes I just use the direct digital because it's supposed to be better, but I've never actually heard that.

But if reference high end products can get away with 91dB SINAD (and far worse at high frequencies), the -120dB added noise from resampling is just not a big deal, and if we still had analog interfaces there could be endless competition, even from people like me.  (I recently found a "open source" surround processor project.  It hasn't gone anywhere in 5 years, probably due to the HDMI issues.)

I had been thinking the Integra Research RDC-7's would give me this kind of analog processing.  For quite some time I was blocked by the fact that they use a snake multichannel input connector, and I never got around to testing the multichannel path because I didn't want to risk getting the wrong kind of snake for it.  Well, it turns out I could have seen the issue from looking at the schematic, which reveals that the multichannel inputs are not digitized, they are simply passed directly to the output with volume adjustments.  This is the same limitation as on my 2005 Yamaha HTR-5790 receiver.  So the glorious Integra Research would not move me one inch forwards, and possibly a few backwards.

I understand that there were some earlier processors by Yamaha and others that did digitize multichannel inputs, at least up to 5.

But to get with the program, and play with all the new surround toys, I'm still tempted to get the AV7705 at a decent price.

TIme Alignment Answer from Linkwitz

Linkwitz explains why I had to delay the signal to the subwoofers to get time alignment with the panels.  At the end of this incredibly informative page on Crossovers.

The woofer's natural highpass behavior causes a phase lead which is probably far from zero at the crossover point and therefore affects the addition of the woofer and midrange outputs.  This can be corrected by placing a first order allpass in the midrange channel which simulates the highpass phase shift of the woofer.

I don't think I like his solution better than mine (I delay the woofer channel instead of phase shifting the midrange channel) but his analysis is crystal clear despite being totally counter-intuitive.

One normally doesn't think of woofers as being a high-pass device, especially subwoofers.  They are supposed to go lower than we can hear after all.  But lower than we can hear, is still infinitely higher than DC.  At some point above DC the woofer HAS TO cut off (except for fan-type subwoofers, and they require massive amounts of power and cannot be made cheaply).

AND, here's the rub.  The lower the low cutoff, the greater the phase lead!  So, woofers will naturally lead higher frequency drivers, since the former have lower low cutoff.

My solution of delaying the subwoofer channel can work perfectly at a single frequency, but not other frequencies because the problem may not be that the woofer is simply delayed...it has a highpass phase lead, which means the delay varies with frequency.  In that sense, Linkwitz' solution is better, it brings the woofer and midrange into alignment at all frequencies.  Mine might be  better because I'm not introducing more allpass behavior in the midrange.

With a reasonably steep crossover, I don't believe the mismatch in allpass behavior is such a problem generally (at least in digitally crossed over systems like mine that can easily apply delay instead), and especially in my system where the "midrange" panels can go nearly as low as the subwoofers.  However, it's worth keeping in mind.  It's an incredibly important insight.  It seems to suggest that my friend who was often criticizing speakers with recessed woofers especially as not being "time aligned" may have been all wrong.  Also explains why one of the least costly "transient correct" speakers of all time, the Spica's, have deeply recessed woofer.  I vaguely recall, back in the day, that a defender of Spica or some such mentioned the woofer lead issue, but I never understood it before at all.  I thought it was the crossover that was causing the woofer to "lead," and in that case the distances still needed to be equal so the crossover would work correctly.  But the problem is, it's not the crossover that causes the woofer to lead, it's the difference between the woofer and the midrange drivers in themselves, their differing low frequency cutoffs.  Ideally both should have identical low frequency delays, but they don't.

However it still troubles me that this "lead" might not be applicable entirely to transients.  That's where my intuitions regarding how filters work seriously breaks down.

It seems to me that information cannot travel through a system in negative time, and that information does not travel through woofers hugely faster than midranges.  Any kind of "phase lead" must represent some kind of information loss, like the front part getting lopped off.  But in saying that, I still can't see how this would apply to the woofer/midrange situation.

Perhaps it's because I'm not thinking of the information loss to the midrange--it is losing the lowest frequency information (because of it's own highpass behavior at least) relative to the woofer.  The lopping off of this low frequency information is possibly causing it to be delayed MORE.  The woofer is being delayed LESS because it is not losing as much low frequency information.

I don't really know how to understand this yet, but it does seem that to work properly in a system with a crossover, the woofer must be delayed even if this means it doesn't start contributing information to the listening position until a later time than the midrange.  And that is what one sees in a decently engineered but nevertheless allpass speaker impulse...in which the high response starts and rolls into the lower responses.  But how would it work with transient perfection?  Perhaps the greater phase correction is needed for the woofer itself than the crossovers.

It seems that electrical engineering has many different abstractions, and issues arise with mixing the abstractions the wrong ways.

Although we are talking about the low pass function of a woofer (Kellogg-Rice dynamic woofer), but actually, as far as the abstract high pass function nature of it, we could just as well be talking about a capacitor in a series-capacitor followed by shunt resistor circuit (following some AC generator), which is another high pass function if we idealize it slightly.

In that circuit, current must lead voltage in the capacitor.  For abstraction purposes, we assume the resistor has no parasitic capacitances and inductances, and as such it samples the voltage at its terminal instantaneously.  (Somehow when people talk about the capacitor in the voltage leading thing, they rarely talk about the load.)

But what does this mean for information traveling from the generator to the load?  I'm thinking out loud here.

At the moment the generator applies a voltage, that voltage appears across the capacitor and the load. But the capacitor is having none of it.  Instead, it all appears across the load.  As current flows through the load, it flows into the capacitor and therefore a voltage appears across the capacitor AFTER it has appeared across the load.  The capacitor is soaking up the signal AFTER it has started. So in the long run, DC or a Step would be quenched all into the capacitor and none into the load.

In what sense, then, is the signal "leading" into the resistor?  In the sense that the maximum rate of rise for the load is at 0 degrees, whereas the maximum rise in the AC signal itself is at 45 degrees.  As soon as the input signal has started, the rate of change in the load begins falling, as would happen at 45 degrees.

The voltage at the load...which is the output of the low pass filter, is therefore 45 degrees ahead of the input signal.

However, none of this means that start time of signal is changed.  It simply starts at 45 degrees rather than 0.  One may be tempted to draw sine waves like arches but they aren't.  The first 45 degrees is a gradually inflecting upwards, starting like nothing.  OTOH, the output of a high pass filter starts at maximum upwards movement.

Now, for a larger capacitor this leading time is longer.  So the high pass filter retains the maximum upwards characteristic longer.  I still find it baffling that this means the larger capacitor "leads" more. It seems to me more that it holds on to the lead longer.

CIVIL: Capacitor: I leads V, and by 90 degrees, Inductor: Lags.

The Capacitor is maximally charging at 0 degrees into the input signal, which means the current is at maximum at that point, which means the voltage across the load leads the AC input signal by 90 degrees, because the voltage across a resistive load has the same phase as the current flowing through it, and because the current flowing through the capacitor must be the same as the current flowing through the load.

This is not to say it's starting any earlier in response to any signal, but that it's starting at the 90 degree point instead of zero for AC waveforms.

I'm still not clear why the lower high pass filter would lead more.  However, if the lower high pass filter leads by 90 degrees of it's cutoff frequency, that would be "more leading" time.  I just don't see how it's cutoff frequency comes into play here.  It should be just 90 degrees.  The capacitor is maximally charging at 0 degrees, i.e. when the signal starts.

And even then, it's still even more baffling how the lower high pass filter leads more then the higher high pass.

The "more leading time" seems to be the correct reason and answer.  The leading DOES have to do with the cutoff of the RC circuit.  It leads by 90 degrees of that cutoff frequency, even if that frequency is no where in the input, that same effective leading is applied to everything along with the associated cutoff.  It's weird but I think I'm beginning to get it.

One again this has little or nothing to do with how fast the input starts, but the phase angles are going to contaminate the step response unless the leading is accounted for...this tends to mean delaying the "more leading" woofer...and it does seem desirable that it should be delayed, and also so that the impulse response "rolls into " the bass, rather than the bass in any way contaminating the leading edge of the impulse.  In fact that contamination is what happens is the woofer is not delayed, as the speaker with the lower highpass cutoff is going to have pre-mature phase angle from a dirac impulse--actually beginning to tend downward as the midrange is still going up, causing cancellation of the leading edge.

The exception would be designing each range of speaker for identical highpass response.  Some very quirky designs do that, not that I'm endorsing them in total, but it's easy to understand the desirability of it now.

The June 2019 TIme Alignment

Finally, as promised (but similar to a long line of unmet promises), here is the report on the living room system time alignment and adjustment near the beginning of June 2019 when I finally turned REW onto the living room system in order to do a measured time alignment which hadn't systematically been done since about 2014...something like forever for this ever changing system.

In May I obtained a Focusrite 2i4 interface so I could do full loopback with REW.  I had been unable to get my historic Emu 0404 to work on Windows 10, so I just decided to get the cheap interface everyone uses now (and which apparently still works on Windows 10, with the status of other interfaces such as the Lynx Hilo uncertain).  I already had a suitable calibrated mike I bought specifically for REW a few years ago.  You cannot use a USB mike during loopback because you must use the same audio interface for both input and output, and a USB mike only has an input interface.

When you do full loopback, in theory, you can simply have REW compute the delay time from the loopback electrical output to the measured acoustical stimulus.  Then, in theory, you can simply dial in the required compensation into your digital processing units.

Using a loopback is also preferred for getting a more accurate computed step response.  REW uses a stimulus which is a sine wave sweep from a low frequency to a high frequency.  The actual response to that is hard to interpret for phase response by eye.  So then REW analyzes the response to this sweep stimulus using FFT, and then it can compute what the response would be either to a dirac type impuse (the "Impulse" response), and a step type stimulus (the "Step" response).  Because it is the simplest possible stimulus, and arguably most informative, speaker designers and reviewers mostly evaluate the step response.  It should always be remembered, however, that REW did not actually measure this stimulus as such, it merely computed that, given the way the unit under test responded to the sweep, this is the way it would have responsed to an actual step.  The computed step response will have artifacts.  But if you were measuring an actual step, it would be highly random because of noise.  The swept sine stimulus, having more broadspectrum (across desired band) energy increases the S/N over what you could measure with an actual Step stimulus.

I had long done such compensation simply based on the physical distances of the speakers as seen from the listening position.  Roughly speaking, the supertweeters are about 1 foot back from the panels, and the subs are about 2 feet back.  Assuming a foot of distance to have about 1 ms delay (I'll use that approximation a lot in this post), I could then compensate by having NO DELAY (0 ms) dialed in to the subwoofer crossover unit, 1 ms delay dialed in to the supertweeter crossover unit, and 2ms dialed in for the panels.   In practice, I intentionally add 10ms to these numbers to facilitate more on-the-fly experiments and adjustments, since you can't further reduce a delay already at 0 ms.  So, a typical adjustment based on distances might be: 10ms delay on panels, 9ms delay on supertweeters, and 8ms delay on the subs.  (In practice, I tried to make the numbers accurate to 0.1ms by measuring to the nearest inch.)

But this is all approximations until you actually do acoustical measurements, especially when all speakers are nothing like point sources.  Real acoustical measurement, not distance approximation, is what has to be done to do the time alignment correctly.  But it's so difficult, and fraught with problems, I haven't actually done it very much.  In principle, I should do a new time alignment after any adjustment of any kind.

It was failure to do actual acoustical measurements (which many subjectophiles and speaker amateurs HAVE NEVER DONE) that led to a huge mistake which I continued for several years!  I started using different DACs on each way, without realizing that the DACs themselves have delay differences, and these differences change at each different sampling rate.  Once I discovered this issue, I quickly gave up on dialing in new compensation when the sampling rate changed.  Too much work.  But that was when I decided to add 10ms extra delay to each path.  However, it turned out that some DACs could add far more than 10ms delay for some sampling rates.  I ultimately decided the best (?) way to handle that variance is to eliminate it by using identical DACs on each way.  I went with inexpensive but decent measuring and pure sounding Emotive Stealth DC-1's (now discontinued) because they had all the needed features: AES digital input, balanced and unbalanced outputs, exact level adjustment.  Audio Science Review puts this DAC down (somewhat unfairly IMO) however if I had to make the choice now, I'd go with the similarly inexpensive DAC they recommend, which has even lower distortion and noise.

This time around, my plan on simply dialing in the REW computed delay for each way was swiftly defeated by an apparent fault with the way REW computes delay for my subwoofers.  I have written a long post about this previously.  I first noticed that the delay times being computed by REW for the subwoofers were about 3ms longer than I had expected.  Then, looking at the computed step response, it appeared that REW was computing the zero point as the beginning of the second hump in the impulse response.  This appears to be an error in REW--while the second hump is slightly larger than the first, it is nearly the same height, and does not appear to be a digital or noise artifact (though there are lots of those, so it's not certain either).  It may have something to do with peculiarities in my system--the rear wall reflection is my current best theory.  This is an example of the computed step response for the Right subwoofer.  The red line shows where REW believes the actual step response is beginning:

It shocked me to see that I got this same result even with No Crossover, which I finally realized I could safely do simply by restricting the range of the sweep to below 300 Hz.  In tests a few years back I decided it might not be safe to send a full range signal--especially a dirac--to the subwoofer with no internal or external crossover being applied.  So I feared sending audio to the sub without crossover.  But this is safe when the input stimulus itself is restricted.  Coincidentally REW defaults to a range below 250 Hz, making it safe to run on most subs without crossover, until further adjustments by the user.  Here is the computed step response with No Crossover:

So, the smaller leading hump which REW's delay calculation ignores has nothing to do with the use of a crossover, which had been the leading theory until I took that measurement (and it's still hard to get out of my mind, because I remembered either lowpass or highpass was supposed to LEAD...but actually it's the highpass that's supposed to LEAD).

Along with the double hump is the extremely curious fact that the beginning of the first hump actually has less delay than the output of the panels, even though the subs are further back.  The beginning of the second hump, where REW seems to think the step response begins, has about 3ms more delay than would be guess from the physical location of the speakers.

Now, REW seemed to give me two choices: go with the officially determined "delay time" which put the sub about 6 feet back from the panels instead of the approximate 3 foot distance, or go with the time of the beginning of the first hump, which put the subs ahead of the panels by about a foot (???)

I was so frustrated by this, I decided to look at something else: the notch in the panel response around 225 Hz which I had long assumed to result from rear reflection/cancellation.  I discovered that moving the speakers further out from the wall could change this a lot, seeming to almost eliminate the problem, and I decided to optimize THIS before doing the actual time alignment.  As I changing the Acoustat position, and looking only at the frequency response btw,  I simply left the delay compensation as I had previously set it by distance approximation.  Starting from the "original" position from a few years ago, which had the 225 Hz suckout, I was getting a step response like this:

Notice the little peak at just over the top of a large hump.  That little peak is the beginning of the Acoustat step response, wheras the hump is the first hump of the subwoofer step response.

When I finally decided to move the Acoustats exactly 6 inches further out (for a total of about 39 inches from the wall) because it removed the suckout at 225Hz in the frequency response I was looking at, the step response looked like this:

You can see the little peak of the Acoustat step response moving forwards in time, as the Acoustats are being moved closer to the listening position, just about to the highest point of the first subwoofer hump.  But even moved 6 inches forwards (which is about 5 inches closer to the listening position) the subwoofer response still seems to start first.  Notice you can also see the little wiggle where the supertweeter is starting about 1 ms later than the Acoustats.  At this point I had not yet moved the supertweeters forward to match the Acoustats, which I couldn't completely do anyway, and it appears their initial adjustment was slightly wrong also, but less than 0.5 ms as you can see in the earlier graph where the supertweeter start is a bit more subtle.

And now, for something completely different, here is what the Acoustat step response looks like just by itself (with the low and high pass crossovers I am using):

It pretty much starts with a spike upwards, as it should.  Somehow the amplitude appears much greater than it does in combination with the woofer response, partly that's due to normalization and the curious nature of step response computation.

I measured and stared at the subwoofer step response for a long time, and finally made a wild guess about where the beginning of the woofer step should have been computed by REW.  I then used the computed panel delay, and MY computed subwoofer delay, to figure out a correct delay compensation.  Sadly I was not taking notes about any of this, it was just a guess, and I tried it, and this combined Acoustat and Sub alignment is about good as anything I've done since, which have only been minor excursions away from this point anyway:

The Acoustat step now merges with the woofer step so that the intial rise is a tall spike, as it should be.  It still possibly looks like there is some preceding bass response, but that's down by at least 40dB, and is probably just background noise and artifact.  I could not make the step look any better by delaying the bass much more than the above.   I tried adding a bit more delay, and the step got worse:

OK, so whatever that was, it looks like too much subwoofer delay has pushed the initial Acousat spike down into the hump in some bass noise valley that precedes the actual bass response, which you can see kicking in a few ms after the initial spike with a blast.

Comparing these pictures begins to get subjective again, I admit, and I wish I had something more solid to latch on to.  It did seem to me that I could try to maximize the length of the leading edge of the spike.  That would indicate that all drive units are contributing maximally to that leading edge.  Now, since REW always places the top of the spike at 0dB (btw, I have not yet calibrated the Focusrite interface itself) maximizing the length of the leading edge curiously means that the leading edge should "start" as low as possible.  You can see that operating in the above two graphs.  In the first graph, what I am calling the noise floor joins the leading spike at -46dB, with the top of the spike at -7dB, so I would count this as showing a leading edge of 39dB.  In the second graph, the noise floor joins the leading spike at -30dB, so the leading edge of only 23dB, clearly inferior.

What isn't so easy to interpret, however, is the the negative spike excursion below the joining point means.  I think it is mostly artifact, but also should be as low as possible indicating complete cancellation or some such.  Anyway, as the delay is adjusted up and down the distance downward changes and also a "gap" seems to open up between the preceding bass-hump noisefloor and the leading edge of the step, making even more a subjective call each time.  As I can't remember the actual adjustment for each measurement (my poor note taking on display again) I'm just going to show a bunch of them, in the order I took them, and where I stopped and why.

Trial 3 has leading edge 24dB, very poor.

Trial 4 has leading edge 30dB (but look at how low it goes also--actually to the bottom of the full scale not shown).

Trial 5 has leading edge 28dB, but deep spike again.

Trial 6 has leading edge 47dB and deep valley beforehand, best so far.

Trial 7 has leading edge 32dB, very poor.

Trial 8 has leading edge leading edge 31dB, very poor.

Stepping back in opposite direction now, Trial 9 has leading edge about 35dB.

51dB !!! This appears to be the winner, especially if you take the beginning of the initial spike at -58dB, then the initial spike is 51dB.  The supertweeter is already being added in and that might account for some of the wiggle at the start--it appears pretty well aligned too.  Alternatively, you could interpret the beginning of the spike as high as -48dB, in which case this is not the winner, but close.

With hand wave you could almost say this is almost looking like a pretty good step response, with a tall initial spike and pretty well filled in right after that.

This is clearly not as good, a step backwards, with the leading edge at no more than 40dB.

After the 11th measurement, I believe I went back to the preceding one, Alignment 10, as the best.  I can't be sure of this, because I didn't take notes, so it might have been one of the other pretty good ones, such as The First Guess.  But I'm pretty sure it was the 10th, because I could still remember the values for the 10th when I did the 11th, and the 11th was clearly a step backwards.

The curious delays in the DSP boxes to achieve the best alignment are:

Subwoofer: 6.3 ms delay
Supertweeter: 4.6 ms delay
Panels: 5.3 ms delay

So the subwoofers are delayed 1 ms MORE THAN than panels, though they are also about 2 feet further back.  That is the second issue I don't understand.

I have already ruled out the lowpass crossover as causing this.  It also probably not because of delays in the complex interface box of the Acoustats, because the delays required to align the supertweeters and the panels is about what one would expect based on distance.

This should not be confused with the first issue I don't understand, which is why REW is apparently computing the subwoofer delay time incorrectly.  This is an entirely separate issue, as I did my alignment by the graphical means shown above, and NOT by using REW's delay estimates, especially for the subwoofer.  However, there might be some connection I cannot yet fathom between the two.  Or maybe not.

Now some might find it outlandish to write so much and not say a word about how it sounds.  Of course it sounds wonderful!  My "electrostatic" bass is now even more well integrated into the sound, without being the least bit less impactful.  Bass lines are easier to follow.  And in many other ways it sounds better overall.

But you should expect that I would feel this way, and it's probably not best to fully trust the audio judgements of the audio investigator, and probably not the measurements either.

Next month it could be different.

Weird Woofer Step Response in REW

I noticed and mentioned this strange problem over a week ago, after I had decided how to do the 3 way system time alignment on the living room system.  I do not trust the Delay times computed by the Room EQ Wizard (REW) program for the subwoofer output.  I found a different way of doing the time alignment using the summed response, but more about that in a future post.  In this post I'm just going to show the problem with the REW computed delay time, which is a very serious issue in my opinion.  I can imagine many people using the delay times to time align their system, as I almost did, and it would be wrong, at least if it's like mine.  (I don't really know what is causing these weird results, perhaps it's something that mainly applies only to me.  As I am writing this post I've completed about six different experiments to try to get REW to calculate the subwoofer delay correctly, but none have worked so far.)

Examining the (computed) step response REW is showing for the subwoofer shows the problem with REW's delay computation.

Notice how the step response begins (that's where it clearly rises above the noise) with a series of humps.  The first hump is not as high as the second, and the second one is the tallest.  FWIW, the black line, which represents the Schroeder Integral, seems to start pretty much where I would consider the bass step to have begun.  But that is not where the computed delay time would lead you.

The computed delay time, which doesn't show up in this graph for no good reason, but rather in an information box for "the measurement" in the REW program, is 12.05 ms +/- 0.042 relative to loopback.  (I am taking loopback at the line loop output of the subwoofer, so it is seeing the same signal as the subwoofer amplifier.  The crossover inside the subwoofer is turned off, so this signal has already been low-passed with 24dB/octave Linkwitz Riley.  I have tried taking full range loopback and it makes no difference.)

If you find 12.05ms on the graph, it is within the rise of the second larger hump, not the first hump.

Now the impulse shown above had a long line of predecessor measurements, which looked basically identical.  I first did loopback using the full range output of my preamp, where it's the easiest to tap off the loopback for my 3-way system.  If I take the loopback at the speaker inputs of the Acoustats, it doesn't include the bass or super tweeter signals--that didn't work at all when measuring the subwoofer by itself.

So then I got more sophisticated, and took the loopback at the subwoofer itself, which has a plate amplifier.  This plate amplifier conveniently already has a balanced "loop output" which I routed bact to my Focusrite interface (it required the attenuator being straight up to not clip) which is also handling the microphone input and stimulus output.

Well then I suspected that the double hump might represent the EQ, of which I am applying 8 additional typically very narrow band and up to -12dB notch corrections to the subwoofer room response.

Not wanting to erase my evolved EQ settings by mistake, I waited until I had time to photograph all the settings in my 3 Behringer units, THEN I saved them to memory (not wanted to do that--which might cause overwriting the current settings by mistakenly selecting "load" rather than "save" until I took the pictures first), then I turned off all the EQ's except for the two HC's which represent the linkwitz riley 24dB crossover.  Then I power cycled the Behringer DEQ (because sometimes it doesn't seem to immediately respond to changes in HC or LC filters, until power cycled) with the subwoofer turned off, then I turned the subs back on.  That's what the picture above represents: No EQ except the crossover itself.

Disabling HC eq's in the Behringer is tricky.  It doesn't want to turn HC or LC filters off when you simply turn them off with the controls.  The "cut" EQ's are "sticky" and remain turned on.  I tried power cycling and even that didn't always work either.  This led to several days of confusing measurements.

Finally I figured out how to shut the EQ's off, AND how to verify the number of HC filters you have enabled (both are important!).

The verification is done by using the REW "generator" function to play a fixed 200 Hz sinewave signal at a convenient level, not too loud.  I set it to -16dB.  The actual subwoofer should be turned off, but even if it isn't, -16dB isn't bad.

The most foolproof way to ensure there are no HC filters enabled is to use the Memory function to reload the "Initial Data" of the EQ.

After that is done, I go to the Level display, choosing in particular the bargraph which has the greatest resolution.  With no EQ active, the 200 Hz tone output level (analog or digital) should be the same dB level, within a few 0.1dB's, of the input level.

When one HC at 100 Hz is enabled, the level at 200 Hz drops about 12dB.  When 2 HC's are enabled, the level at 200 Hz drops about 24dB.

Then, the second HC can usually be disabled by turning it off, then pressing the "reset EQ" button in the PEQ second page.  If that doesn't work, power cycling might help.

When there is only one HC enabled, turning it off and pressing "reset EQ" doesn't seem to work.  In that situation, the working solution is to reload the initial data again.

Until I developed these strategies, I was beginning to fear that that two HC's couldn't be run at the same time because the result did not seem to be a steeper slope than just one.  (It turned out, I was not correctly turning off the second one, so I was comparing 2xHC with 2xHC.)  Then I feared the subwoofer itself (Ultra PB13) had serious rolloff above 100 Hz even with no HC filters (and the crossover in the subwoofer itself has always been turned off).  That resulted from not correctly getting both HC's turned off, which may require that loading the inital data step I just mentioned.

Anyway, when I finally figured out how to set the crossover correctly, and verify it, with all these variations, the subwoofer step response stayed fairly similar, and in every case the "delay time" seems incorrect in about the same way.

Here's the step response of the woofer measured with 24dB/octave crossover again, this time slightly differently but the same as in the next two graphs.  Because of how I captured this image from the "Impulse" and not the "Filtered IR" tab of REW, the Schroder Integral is not available.  The leftmost red dotted line is showing where the delay ends and the step proper begins (this info is shown in this unfiltered version, but not the "Filtered IR").  It is showing the beginning of the step at the beginning of the second hump, or 12.5ms in this measurement using loopback from the sub loop output.

With the 12dB/octave crossover, the humps look thinner somehow, but the beginning of the step is still being calculated to be at the beginning of the second hump at 12.5 ms.

With NO HC filters (no crossover!) the humps are even narrower, with the first hump turning into two humps, and now the computed delay falls at the beginning of the third hump, at 15.3ms.

In no case does REW compute the delay to begin at the beginning of the first hump, but that's where my other method seems to land for the time alignment with the beginning of the panels and the super tweeters.  The step responses for the panels and supertweeters does not have this issue at all, the computed delay begins right where the first hump rises from the noise.