Living Room System

Monday, August 3, 2015

Calibrating the Super Tweeters

Not in all the 5 years that I have owned the Elac 4pi super tweeters have I set their level by a measurement comparing their output level to that of the Acoustats.  I have set their level by ear (of course) and by overall system response measurements, but those don't give very much detail about the high frequencies where the super tweeter is operating.  I high pass the super tweeters at 20kHz and the Tact RCS spectrum analysis I have most often done has one point at 18kHz and one point at 20kHz and I wonder about how accurate they are since microphone angle has a huge effect on such frequencies and for stereo measurements I point the microphone straight forward.  Also the Tact microphone calibration itself has been called into question.

It is not easy to accurately measure levels at 20kHz and above.  But for the purposes of determining relative levels, it was simple enough to temporarily change the high pass to a lower frequency and test some signal that is easier to hear and/or measure.  I had been thinking about doing just that for several years, and that is what I finally did last Sunday.  I made a 30 minute aiff file with a steady 14080 Hz tone (A9).  I figured this was good to test tweeters: high and away from any low cutoff, and yet still not too high to be dangerous or inaudible to me.

I used Sox to make the file with the command:

sox -n a14k.aiff synth 30:00 sine 14080 sine 14080 gain -20 rate 44100

For some reason that wasn't acceptable to Sonos.  So I ran the file through Triumph, which had no problem re-writing it.  Then it was OK to Sonos.

At 73 dB playback level (6 inches from the Acoustats) it was just barely audible to me.  At about the same distance by eyeball, the Elacs measured about 3dB less, about 70dB, in both speakers, with the DCX crossover set to 10khz.  (The Elacs may also have their own internal high pass at 4kHz.)

It was difficult to do this near field measurement because it is highly dependent on distance, and I don't think right at the surface is valid (though…it might be as good or better than what I did).  I was thinking of attaching a ruler to the Galaxy SPL meter, but then I decided this measurement was good enough.  I was very pleasantly surprised, almost incredulous actually, that I had set the tweeters this close to the level of the Acoustats (and that is a very long story…they've been set all over the place on the DCX and further confounded with the Parasound amp's own attenuator, and a pair of Harrison Labs attenuators that I may have used different versions and sometimes had turned the wrong way).  By this primitive measurent, it seemed likely that I had not been setting the Elacs way too high, as I had been doing at one time and feared I was still doing.

It did occur to me that any near field measurement as I was doing was likely to be unimportant because the Acoustats have far more total radiating area.  By that standard, the Elacs might need to be set not just 3dB higher but 10-20dB higher.  I couldn't really figured that.  I tried measuring from my chair but the it was hard to get a good measurement without cranking the tweeter level much higher and I decided not to do that this time.

For awhile I tried setting the tweeters 4dB higher.  But, just to confound things further, I reset the crossover point down to 18kHz and I'm not quite sure if I had previously used BE24 but somewhere along the line I did that too, now that I was no longer worried that I was overdriving the hell out of the Elacs.

Well this indeed seemed to clarify the Sax in Jesse J's Lovesong, which had been seeming a touch dry on my system as compared with Luther's.  I have often found (or I should say, I though I found in sighted listening tests) that the Supertweeter actually removes harshness and glare, and that seemed to be the case here too, at first.  But then I did find it overbearing, and dialed the super tweeters back to 2 dB higher than the previous level (see below).

Well after this I couldn't stop playing with the adjustments during each thing played because it didn't sound or figure right one way or another.  One thing for sure, with the crossover reduced to 18kHz, the level raised 1-4dB, and the slope changed to BE24 or LR24, the output of the super tweeters has become clearly audible near the speakers themselves, which wasn't necessarily true before.

And it seemed like I was hearing huge differences with the slope changes.  BE24 did have a clarifying intensity, but also seemed to add some roughness (possibly boosting the 6kHz region, or causing comb filtering at more audible frequencies).  It was instructive to listen to the tweeters without the Acoustats playing, then the differences in the sounds of each option was far clearer.  The LR crossovers sounded by far the most refined, with LR48 the most refined of all.  I believe I had previously been using LR24, which reduces the super tweeter output to nearly nothing (but what there is is high high high).  But with LR48 at 18kHz, the effect sounded very top heavy.  The BE24 and LR24 seemed to have the least top heavy sound, a natural balance.  I ended up with the LR24 which sounded good by itself and also blended well with the Acoustats giving a very clear yet refined sound without coarseness.  The current adjustments as I am writing this:

19.2 kHz
6dB attenuator

Note that this was -2.3dB by the old standard where I was boosting the DEQ for the panels by 1dB.  That seemed to work weirdly I determined.  With 1dB of boost, an input of -2.5dB caused peak clipping at the digital output.  I dialed back the boost to allow higher levels (and Monday I have been using +3dB on the Tact (96.9) when playing SACD from the Denon which peaks at -4dB).  I have to make sure I'm not loosing any digital gain because I need pretty much full digital level out of my DAC's for a good loudness.

(The difference between the output of the Onkyo which now powers the Aragon, and the DCX which powers the Parasound, is on the order of 12dB.  Oh, and the Acoustat 1+1 are about the most inefficient speaker ever, in the 70's!  Whereas the Elac speakers my tweeters were matched to were in the upper 80's.  So actually a combined attenuation of only 9.3dB seems to small, intuitively I would have expected 24dB or far more, and before these tests I wondered if I wasn't setting the level way too large and causing a huge ultrasonic peak.  Update: now I wonder if the Elac crossover was at 14k, and my measurements off by 6dB or more as a result of that--in which case I now have a slightly larger ultrasonic peak than before, but in which case it must not have been as large as I feared anyway.  I need to retest with higher frequencies.  And I should record amplifier drive levels and push the levels harder now that I know the Elacs can officially handle 400W continuous and 600W peaks, or at least a later one did.  So I should be able to reach above 80dB and make proper listening seat measurements.)

So when dialing back the DEQ boost to 0, I reduced all the DCX settings to compensate, putting the tweeters at -3.3 from -2.3.  When the day had started it had been set to -4.3.  So this was a 2dB rise as seemed conservative from measured levels, with twice that much sounding too much.

This is almost like having super tweeters for the first time.  You would have thought I'd do this kind of calibration the day they arrived.

The 19.2kHz LR24 seems intuitively consistent with the limitations of the Acoustats FWIW, just where a super tweeter should fill in.  20kHz was probably too high.  I suspect the actual rolloff of the Acoustats is close to 24dB/octave also, including 6dB/octave from the resistor in the crossover, the trannies, and the mass of the speakers.

Also contrary to earlier fears about one being bad from ribbon warping, both super tweeters measured the same in level and sounded the same.  (I've seen some ribbons even more warped than my left tweeter.)

Here are the specifications for the Elac 4PI Plus, which I think was a later model.  Mine is more impressive looking, and I think a very similar earlier model, except it did not extend to frequencies higher than 35kHz as later models did, but possibly extended lower, perhaps to 4kHz.  Well according to these specifications, not for my exact model but similar, the maximum power handling is 400W continuous and 600W peak.  They should be safe with the current Parasound HCA-1000A amplifier, but maybe weren't safe the the Acurus A250 amplifier I was using before which can produce 500W or more into 6 ohms rated impedance.  At least once, due to some technical error, I was clipping the A250 into these speakers at peak.

Other specs show sensitivity variable from 84-92dB at 2.83V (this model had sensitivity and crossover controls, I believe mine has fixed crossover but I'm unsure of the frequency).  The crossover is selectable for 10, 12, and 15 kHz.  Frequency range 10-53 kHz.  Impedance >10k ranges from 8-3.5 ohms.  Weight is 4kg (about the same as mine).

WRT Bessel "filter" (it's actually a class of filters some say) something like that may form the basis of the crossover for the Spica T-50, which has excellent time response but ripple in the frequency response around the crossover according to some.  Because of ripple, it hasn't generally been recommended for crossovers.  The "bessel high pass" does not retain the perfect linear delay that the theoretical low pass does, it's merely the "maximally flat" case given the amount of phase shift which must occur.  Actually since LR24 has Q of 0.5, and BE24 is actually slightly higher, the LR crossover would have more gradual phase shift I would think, but I'm thinking it might be a more constant phase shift.

In my case, given that I am correcting the low pass of my panel speakers, I want a matching time response around crossover to that of my panels.  If the panels roll off like a pair of second order butterworths in cascade, and they probably do something not entirely unlike that because of cascading rolloffs in the transformer primary and secondary circuits, an LR24 or similar high pass on the tweeters would provide the correct inverse time response, and the flatter Bessel filter would be too-flat in the time domain for correct cancellation and cause some comb-like responses.  Perhaps that was the grundge I thought I was hearing.  But I would find it hard to believe I correctly sorted that out by ear, and I still suspect my selection of LR24 was more related to loudness than anything else, it was less loud that the BE24 but more loud that XX48.  Perhaps another crossover function would work better at a different loudness level.  I have not tested that.  But what I am doing seems intuitively correct or close to it, and sounds good.

Fixing the SCD-1, SCD-777ES, or other transports

Advice on fixing SCD-1 or similar transport

General advice on fixing CD players from Lampizator, a celebrated audio modder.  Lampizator has spent more time noodling around with old CD players than anyone, and he says he has only once ever seen a bad laser, and that was caused by his boosting the laser drive signal.  Most of the time, CD problems are caused mechanical failures of various kinds, many due to accumulating grunge or hardening lubricants.   Lamizator curses the professional techs who make bucks by unnecessarily replacing lasers, therefore depleting the precious stock of factory parts.  He says that if you can hear the disc spinning at a controlled speed for 2-3 seconds when the disc is loaded, the laser is fine because in most players the disc won't even start spinning unless the laser detects a reflective surface.

It does now seem that the "fixed laser" transport (FPM) was Sony's most intense, and probably best, effort to remove transport vibration and jitter.  Sony had been making these transports back to the beginning of the CD era with the professional CDA-5000 Compact Disc Analyzer, and a series of high end CD-only models in the 90's also used a fixed laser transport, such as the CDP-XA7ES.  The last go around was for the SCD-1 and SCD-777ES models.

The Vintage Knob says:

FPM was intended to suppress vibrations where there are more in the first place (the spinning CD) rather than try to minimize those which have a much lower magnitude - in the laser pickup itself.
So the disc's spinning motor and rotor become the moving parts (one big moving part), and the laser block remains... fixed on its base.
But TVN also says that FPM was less expensive than Sony's other premium mechanism, the all aluminum BU-1 with magnetic rails.  And lower cost was a key motivation to bring it back in the mid 1990's.

The SCD-1 did achieve a lower jitter measurement in Stereophile than I remember for any other player, FWIW, playing CD's.  However I have not tried doing a systematic comparison and the jitter measurement techniques may have changed over the years.

While the original high end SCD players have many SACD loving fans (the consensus emerged almost right away that there were better Redbook players from Esoteric, Krell, Marantz, and others) many have had a love/hate relationship with the transport.  It is apparently somewhat unreliable, and by 2006 or so many owners were stocking up on lasers, sled motors, and spindle motors.  Sony tends not to offer service or even parts after 10 years or less, and they may have already used up all the spare parts before them in earlier repairs (sometimes unnecessary part replacements, according to Lampizator).

The general consensus now is that Esoteric makes the best mechanisms such as the VRDS which itself comes in several versions.  But if I want a "reference" player specifically for SACD, the SCD-1/SCD-777ES still represent a kind of reference benchmark.

Here are the 3 parts that some people were stocking up on:

1. Spindle motor: P# 1-763-254-11
2. Sled motor: P# X-4952-147-1 
3. Laser Pick up: P# A6062396A

Here's a thread with a guy fixing an SCD-1 which originally had pictures.

A friend has a boom box with cassette and CD player.  The CD player is often not working, though I got it to work a few days ago up to track 7 out of 10.  I think there is possibly something making laser movement difficult.  We agree it is worth fixing this because CD/Cassette boom boxes are basically not being made anymore, thanks to the kind of planned "markets" we have (planned by corporations to extract the maximum from us).

It's a CFD-S26.  Here is a similar model being repaired:

And here's a DIYAudio thread about fixing CD transports that starts (near the end) to discuss the adjustment of the sacred pots "track and gain" and "focus gain" that the service manual advises you not to adjust.  Well apparently lots of people with nothing to loose often try and win, for a few months anyway.  I once saw a friend adjusting those pots…and it looked just crazy.  After watching that you wonder how CD's mostly work at all, if just a hairs breadth of a turn of some pot inside is the difference between Perfect Sound Forever and junk.  But then again, I once calibrated a Sound Technology 1700 analyzer…and that's a similar experience.

And here's a more detained DIYAudio thread on CD player restoration and adjustment.

Sunday, August 2, 2015

Fixing the symptoms first

I heard Jesse J from Oregon in a live concert Saturday, bought the CD, got it signed by Jesse, played it at home, was wonderful.  Though I though a bit more wonderful when I lowered the TacT level (92.9=0dB) to 79.1 on cut three, that may have reduced some critical rattling in my living room.  This CD is no wimp when it comes to deep bass!   The next day, my friend thought it sounded a bit mechanical.  Right at that time, there were several rattles obvious to me, I've set about fixing them, which brings me to the title point.

As a general rule in such situations, it's better to fix the symptoms first.  By symptoms I mean the effects farthest from the original causation.  Why?  Because if you wait until after fixing the primary cause, you might not be able to find the symptoms again.  So I will fix the rattling before fixing the frequency response.  The frequency response doesn't require a non-linear failure like rattling to measure and can be fixed after the rattling is fixed.

But actually the frequency response is not bad.   I went through the test frequencies I have used to make adjustments before.  I have a series of tones recorded at something like -20dB, in Hz:  16,18,20,22,25,28,31.5,36,40,45,50 and up to 160.

Sitting at the Kitchen table listening casually, it seemed like 36Hz and 40Hz seemed a bit low.  But measured at the listing position, 22Hz-45Hz are within 1dB, which is pretty incredible and testimony to my hand-tuning of the EQ's, especially to fight the 10dB room mode at 45 Hz.

Playing the subwoofers alone makes it easy to find the rattles.  The bass is amazing tuneful considering the 80 Hz lowpass and everything.  I did think there was a kind of excessive bounce, excessively electronic sound in the bass, which I think is an unfortunate attribute of my system (I am not faulting the recording, which sounded just fine and natural on the bass on another audiophile's system which doesn't go quite as deep).  So, for that reason, and also because of booming and rattling all over, I dialed back a 20 Hz boost I had in one channel to zero.  Now there is no boosting below 31.5 Hz (at one time I had several boosts below 31.5Hz to enhance the lowest bass) and in fact there was a -3dB Q=1 cut at 20Hz, which I dialed back to -2dB after removing the 1dB Q=5 boost.  I can see the trick I was playing here (high Q boost combined with lower Q cut) but it's precisely that sort of tricky EQ that could add an artificial sound from extended reverberation.  Anyway, it's hardly any different from a 1dB change overall.

I only measured 20Hz after removing the 1dB boost, and it was actually 2-3dB below the 22-45Hz level.  But 20Hz is quite problematic, it causes a lot of wall flexing and rattles the kitchen range, among other things, which I can't fix.

The rattling wall clock was fixed by applying self-stick felt to the back, top and bottom.  Now this had an unintended effect of forcing the bottom of the clock away from the wall because of constraining the space for the wall hook on top.  At first it appeared as though the bottom was entirely away from the wall, but after some time, one side is almost touching the wall, the other side about 1/3 inch away.  I pound on the wall and it doesn't seem to rattle at all, so this is fixed for now, but may need to be monitored.

The rattling china cabinet was investigated, and finally the source of rattling was found: two vessels on the bottom shelf were touching.  That was it!  I was prepared to use felt, whatever, but that wasn't necessary.  Going one step further, I pounded on the shelves with my hand, and found one other potential rattle in the teapot and butter dish, with both have top and bottom parts.  I wedged pieces of paper in between top and bottom pieces and now they seem quiet.

The rattling in the range hood and back plate has eluded me for now.  Nothing I could hold with my hand changed it.  Anyway, this was not the rattle I heard on Jesse J's album, but something else that turned up playing spot frequencies up to 100 Hz.

The "cause" of lack of LF absorption will be a long time in fixing.

Friday, July 31, 2015

The Great Debate: Amir vs Arny

The Great Debate, Part 1110250, at AVSForum

The Great Debate, Part 1110251, at What is Best Forum (where Amir is a moderator).  Amir starts it by saying this thread was inspired by a thread at AVSForum, although he didn't link to it.*  The previous thread seemed to be the one…but I'm not sure because there were many others!  And, note that the threads I have linked are a hundred pages or more.  Are audiophiles obsessed with their arguments or what?  We are (yes me included) apparently argumentophiles also.

(*Sadly, typical of how Amir argues, without linking back to demonstrations of crucial points.  In contrast Arny's arguments are almost always backed up with a plethora of links, though not always of the highest quality and many are now dead, including his sadly long gone website pcabx.  For the service he has rendered to society, yes simply by being a gadfly and contrarian to the mainstream high end audio industry, I think Arny should have the resources of a titan of industry himself, but I fear it's not like that at all...)

Technically these debates are about whether the potential improvements from "high resolution digital audio" (meaning it has greater than 16 bit resolution and/or greater sampling rate than 44.1kHz).  But it's also a debate about methods, results, and personalities in all aspects of The Great Debate between White Hats and Black Hats (the terms coined by Peter Aczel, a self declared White Hat who believes that most decent amplifiers sound the same, the classic White Hat position going back to the 1970's).  We've apparently even superseded the always shaky term "Audio Objectivist" (having nothing to do with, say, Ayn Rand) and "Audio Subjectivist" with Amir calling himself the Objectivist--so where does that put Arny?  The object/subject difference hardly gets at it either, since Black Hats (and especially Grey Hats, like John Atkinson--I respect him and his self identification, and myself) may be quite into certain kinds of measurements, or at least technical phenomena, real things happening, that just don't happen to tickle the White Hat sense of importance.  And a hidden argument which is never even touched is absolutely fundamental--must everything pass DBT immediately, and what if not?  Should technical criteria (say, Jitter) be ignored simply because there is no current DBT proof of their importance?  I strongly think not, however any person not an Amir or higher titan of industry has to prioritize, and I've been shaming myself for spending several years now far more on digital issues even at the fringe of the Grey Hat regime, when I sort of intended to have been working on room acoustic issues instead.  So Jitter should not be ignored, and let's have John Atkinson's and even better jitter tests developed.  But generally, we should probably move on to more important things.  We should be skeptical of all existing Black Hat claims, but not necessarily at the existence of to-be-found issues discovered by Black Hats, even as White Hats do, though the White Hats always often end up with the more minimal explanation.  So the world needs everyone, even as it needs everyone's mind to evolve.

Amir Majidimehr (principal of Madrona, formerly VP of Windows Digital Media at Microsoft) and Arny Kruger have been frequently sparring about what equipment differences are actually audible.  And this goes way back.  Though I'm not sure how long Amir has been involved in these debates, Arny has been involved since at least 1976, and he has recounted a history of the development of the ABX test method around that time and the ABX comparator in 1982.  He was one of the principles of the ABX company.  The very creation of the ABX test method over 30 years ago was precisely to answer the question of what equipment differences are audible.  Way back in the mid 1970's as an audio society member Arny was involved in these debates, and he's still at it now on a multitude of websites.

Reading Arny (who I've never met in person) I am very impressed with his arguments and generally the ways he makes them.  I'm also very impressed with his patience and dedication.  I think I take his side, mostly.

Reading Amir, it is clear he is very smart and is a very technically qualified professional audio designer.  I believe he is honest and not a shill.  However I think he makes poor and sometimes ugly arguments (often to authority, and sometimes to his own authority, and often discrediting the authority of others) far more often than Arny does.  He also seems to me much more to be a tireless bully.  Nevertheless, Amir may be right about some things.

Sadly, in these arguments, there isn't really a suitable technical qualification.  Certainly being an electrical engineer, as such, doesn't necessarily make you an audio scientist, and The Great Debate is not engineering it is Science.  Within Science "Audio Science" is too small to be particularly tractable.

The Audio Engineering Society (AES) is really an engineering organization, not a scientific one, but it does strive mightily (and perhaps too mightily) to retain respectability.  Therefore it is not surprising it cleaves tightly to Double Blind Testing results in its papers which are often written by academic scientists and not engineers.  Meanwhile, many audio engineers don't bother with DBT's.  Many have never done DBT's and never will, but nevertheless often are believed to speak with authority about such matters.

The truth is, right now, there is no authority.  And it is difficult to establish one given all the possible economic conflicts of interest, not to mention egos, etc.

Amir starts the second thread with a post showing a DBT result which confirms his ability to hear the benefit of high resolution.  At the beginning, Arny was not posting (and in fact Amir said that Arny had permanently retired from posting after Amir had posted some brilliant refutation--another unfortunate Amirism).  I have been unable to confirm that Arny ever quit posting anywhere, and in fact Arny posted to this exact thread some time later, as well as continuing to post at AVSForum and HydrogenAudio).

It happens I have seen at least one of Arny's argument which has has often made with regards to some DBT results.  He has argued that high frequency nonlinearity in amplifiers, speakers, or headphones can produce differences at audible frequencies, and that is what people are hearing.

As it turns out, at least in the first page or so of the second thread, Amir did not reveal the particular equipment he had gotten his positive test results with.  I had read up to the first point and which that question was asked an an answer still was not provided.

Now quite often subjectivist reviewers are quite clear about what equipment they have used to perform some test, and when they do so they go into great detail about every last cable involved, because of course they believe it is all of importance.

So it is more than a bit suspicious actually that Amir left out this detail.

I have not read all of either thread, though it looks somewhat worthwhile for someone like me who remains very interested in The Debate, despite going on for hundreds of pages.

One of the high points of the second thread is where Amir presents a very respectable paper published by AES (Convention Paper 9174 presented at the 137th convention of AES in 2014, by Helen Jackson, Michael Capp, and J. Robert Stuart) recently proving the audible differences of different kinds of digital filters.  That result does cleave very much away from the "all digital sounds perfect" position of the White Hats, including Arny.

(Similar experiments in the past with positive results have been shot down on the basis of sampling artifacts caused by equipment configuration.  In a earlier blog discussing that, Arny nails the best way of preparing test material.  It should start from the High Res material then be down sampled to the lower rate.  THEN it should be up sampled to the high rate again so as to avoid playback differences caused by switching sampling rates.)

Now one wouldn't think J. Robert Stuart (a Fellow of the Audio Engineering Society) would make such mistakes.  But it seems he may have padded this test in various ways, according to Arny and others at Hyrdogen audio.  It's clear from information available that Stuart used defective Rectangular Dither in this test.  He tries to justify this on the basis that "it is often used."  He knows it is not the best because he makes products that use the better method (some kind of triangular).  The question being addressed is not Rectangular dither.  The correct approach is to use the best available technology except for the item being tested.  So, sadly, this positive looking result is not acceptable and the Convention Paper now looks more like a puff piece for Meridian.  AJ at Hydrogen Audio writes:

The BS test is a complete farce and fabrication of results in a desperate attempt to justify $$$ales of "Hi Rez" which of course nosed dived once people realized the scam, confirmed by M&Ms AES peer reviewed tests of actual audiophools, their hardware and purported "Hi Rez" media, the EXACT conditions they and the scam industry claimed to be able to "hear" differences.
The scam industry does not require the audiophool be "trained", the "Hi Rez" equipment/system to be certified, the room to have a specific noise floor, or the music content be cherry picked and doctored as in the BS test.
And he goes on…  But I believe he meant to say "the music content must be NOT be cherry picked and doctored."  That may be required for some official standards.  But if we are talking about the limits of audibility, cherry picking is fine, and doctoring is fine as long as the best available technology is used except for the item being tested.

I will say emphatically that a person's achievements in audio engineering do not necessarily qualify that person or any person to make an authoritative statement about The Great Debate.  Even very qualified, experienced, and successful audio engineers are not necessarily up to speed on this.  It requires a skeptical stance toward many things, which is not engendered in our society.  Such skepticism is not about making money, and these days the incentive exists to show that everything is audible, because that sells more stuff.  Successful audio engineers can therefore not be expected to have explored it very deeply as part of being successful.  Even what some experienced engineer says is not evidence.  The only admissible evidence is from actual double blind testing done to the highest standard, and every aspect of that testing is open to deep criticism.  Will progress be made?  Who knows!  Heat death of the Sun or the collapse of human civilization may well occur first.

My position remains that the audible differences that Black Hats obsess about and White Hats dismiss are likely very small, if they exist at all.  That idea is tangentially supported by a very sophisticated exploration of the meaning of p values in testing I have been reading:
Getting a big p-value is not, by itself, very informative; even getting a small p-value has uncomfortable ambiguity. My advice would be to always supplement a p-value with a confidence set, which would help you tell apart "I can measure this parameter very precisely, and if it's not exactly 0 then it's at least very small" from "I have no idea what this parameter might be".
OK, this doesn't appear to apply to the Great Debate at all, because it's concerning the situation even when you have small P values, whereas the problem with most DBT's in The Great Debate is the continuing lack of small P values generally.  But turn it around, and you see even if we were getting small P values consistently it still wouldn be The Proof many people want.  The fact that p values are related to effect size and sample size means it's not easy to tease these things apart.  The safest thing to assume when an effect that is expected isn't verified in DBT is that the effect is small, not that it does not exist.

But this is never what the blackest of Black Hats say.  They always trumpet their unproven differences as very important, because otherwise why would anyone spend the megabucks necessary for Black Hat tweaks such as cryogenically treated cables?

We cannot trust the most successful audio designers or retailers any more than we can the most successful lawyers.   We can have much more trust in the White Hats who have gained little and have nothing to sell.  Things are not proven until they agree also.  The whole history of high end audio is full of flimflam, lies, and half-truths.  Progress has been made, but more slowly because of that.

Monday, July 27, 2015

James's Charlie and SAE

The low serial Sumo Charlie with rack handles (indicating final alignment by James Bongiorno) is a remarkably nice sounding tuner.  As with the non-handled Charlie, there is reduction of ambience, but not nearly as much.  The quieting is just as good or better.  It is able to hold the weakest of my favorite stations, KSYM, in wide stereo.  That was just the trick, I thought.  Then I brought back the other Charlie and unlike the week before, it was now holding KSYM in wide stereo also.  Shows the need for quick A/B switching when testing tuners.  Even if you can remember the sound or reception objectively, it can differ from time to time.

If Charlie can't hold KSYM in wide, it has to be switched to the inferior narrow.  Actually I had no need to use the Narrow IF in my one day testing of the handled Charlie, and I forgot to do so, but it would be interesting to see if it sounded better too.

The handled Charlie has the punchy bass and clear highs that I associate with the Marantz 20B, which I think Bongiorno may have worked with when he was at Marantz in the mid 1960's.  I suspect that is where he learned the best way to align an IF, or at least the Marantz 10 and 20 way, and he carried this knowledge with him to SAE and Sumo.

And speaking of the Bongiorno sound, I unpacked the SAE MkVIII to check it out, and it had, fuzzily, a similar sound.  This was the first tuner whose design could be more or less fully credited to Bongiorno.  He had been hired to save the MkVI tuners, but he had to work with what SAE had already done.  On the MkVIII, he was given a free hand to make a cost-reduced tuner nearly as good (and it could actually be better).

Despite my belief that air capacitor front ends are better, I actually though the Charlie to have better sound than the SAE MkVIII.  The SAE was slightly noisier, though, and the punchy bass and clear highs were a tad less punchy and clear, respectively.  Perhaps it needs a front end alignment, which would be possible I think, or a refurb.  An IF alignment isn't possible because it uses potted IF modules.

Both these tuners have issues not revealed by the eBay sellers.  The Charlie has the thing I fear the most.  It has a serious smell problem, like the Marantz 2130 I now have in storage because I couldn't bear the smell inside my house.  A previous owner must have been a heavy cigarette smoker.  In all the equipment I've bought on eBay, I've only seen this problem with 2 different FM tuners.  I will not be able to keep this Charlie permanently in my house, though it seems OK now in the garage room which has the best ventilation.  But it sounds so good, I can't just return it or toss it either.  I have to continue with my Investigation of the Bongiorno-sound tuners!  My ultimate goal would be not just to determine how they differ among themselves and compared with other tuners, but why.  Just how exactly was the Charlie aligned by Bongiorno, and how did that differ from what Sumo did after Bongiorno left?  What did Bongiorno know about the IF alignment of the Marantz 10/20 and are they different fundamentally from other tuners?

The SAE has a problem with its numeric readout.  Not just the broken segment claimed by the seller, but an additional and far more bothersome problem right now.  It doesn't show the station frequencies accurately at all.  It is way, way, way off.  When tuned to 90.1 it shows 80.5, which isn't even a valid FM frequency.  When tuned to 88.3 it shows a number in the 70's.  But then once and awhile the number displayed will briefly snap to the correct number, and then back.

The numeric display on the SAE is utterly unlike that on the Charlie.  The SAE only shows the correct number (or whatever it shows) when you are actually tuned in to a station.  Until you are actually tuned into a new station, it shows the number of the station last tuned in.  So it's not as helpful as you might think.

Sadly the motorized Kenwood KT-413 isn't very good at all, and just as FMTunerInfo says.  But I'll add more.  The motor is very fast, even when switched to the "slow" position, and I think that is part of the problem.  Sometimes it actually skips over stations going one way, but finds it in the other.  One wonders how accurate the tuning is.  Anyway, the sound is noisy and wimpy and utterly unlike the Charlie.

Sunday, July 26, 2015

DAC output level measurements

Since this blog is my notebook, here are the measurements of my various DACs, with 10M DMM loading (sorry didn't bother with different), at about -26dB (whatever level, it was consistent):

Denon DVD-5000:
L: 41.6 mV
R: 41.5 mV

Onkyo RDV-1:
L: 48.8 mV
R: 48.0 mV

Audio GD DAC 19
L: 63.0 mV
R: 63.3 mV

The Onkyo does show the most inter channel difference at 0.14dB, which is just above the usual 0.1dB matching requirement.  The Denon is the most tightly matched at 0.02dB.

I suspect the Denon has comparatively low output because it is saving output range (which may actually peak above 3V) for HDCD boost, but this is just a guess.

This shocked me to calculate, but the Audio GD is 2.2dB (left) and 2.4dB (right) louder than the Onkyo.  I would expect the measurements with actual amplifier loading (22k ohms) to be 0.2dB or less different at most and probably below 0.1dB different.

While ABX testing sets a condition at 0.1dB matching, typical matching isn't that good.  Some preamps with steps won't let you set balance closer than 1dB, though 0.5dB is common on digitally controlled preamps.  Old fashioned potentiometer volume controls might have +/-3dB inter channel mismatching (a 6dB total range)--so you really needed to use the companion balance control.  (I was glad to dump those in my then-main bedroom system in 2005 or so with a Classe CDP-35 preamp with digital volume control in no small part for the sake of being rid of potentiometer balance variation, which I barely remember but when I do it is with great dread.  Even my Aragon 28k preamp had considerable channel variation at the very low position I had to use on it.

Friday, July 24, 2015

Audio Nirvana Speakers…Slam!

I listened to these at a friend's house, Audio Nirvana speakers (12 or 15) in large and tall cabinets, with the speakers just above ear level IIRC.

It was in many ways very impressive sound, dynamic, full of slam, powerful and deep bass, clear midrange and highs, excellent left to right imaging, stable but not too much pinpoint.  He said he was unimpressed by other speakers he had heard, often very expensive, in audiophile's homes.  I sympathized with that, and in some parameters (dynamics and slam, especially) I have only heard megabuck systems at audiophile shows, like the largest MBL demonstration, that actually bettered what he could do.  Even with all my complicated and intentially overwideband system, I actually can't quite do slam as well, maybe, except perhaps if you are sitting in a corner instead of the sweet spot...

But I wouldn't say it was the least colored sound, it had clear colorations, a slightly exaggerated mid bass, elevated mid highs, obviously absent highest highs.  They always say things like "most people can't hear above 15,000 Hz anyway" but it seems to me, if it's missing, I can sense that somehow.  Though I couldn't name any instrument or anything that was obviously missing except for possibly noise anyway.  The other colorations seemed apparent but not particularly annoying either.  Possibly the worst aspects of this speaker could be fixed with digital EQ, making it truly reference, save for the supra 15 kHz…And there, supra 15kHz super tweeters might do the trick, without much messing the actual imaging.  Though once we get into all these further approximations of the needed changes, we would likely end up with…less slam.  The apparent missing high frequencies could also probably be adjusted with speaker aiming (beam that center right at the ears) and replacing damping used in the room with dispersion.  Much of what I was hearing as missing highs might not have actually been the speaker limitations.

Distortion, I don't know, he was playing pretty loudly in a small room, and there didn't appear to be anything like rising distortion near clipping.

But there was not much depth, and I think that was the ultimate sign of a cone speaker distortion type*…which should be expected!  The dynamic speaker is driven from the near center, and the propagation of sound through the speaker cone is dependent on frequency.  And the propagation through the speaker cone creates delayed radiation of that frequency for as long as it takes to travel through the speaker cone to it's ultimate point.  It's very fast in a relatively stiff cone (compared to air, say) but not instantaneous.  So there is an irreducible amount of time-smearing, and there may be some distortion from traveling through the cone also.

(*other explanations are possible, say related to room acoustics and speaker placement.)

There is one single cone (or mostly single cone) speaker which gets around these issues, and that's the Walsh driver, where the contour of the cone exactly matches the delay for time alignment.  And it also does it with omnidirectionality.

Actually, a relatively flat and stiff but still contoured cone like the Audio Nirvana may already do that same sort of thing to a considerable extent.  If we want to understand just this angle, we have to do the kinds of measurements few people bother to publish.  The sound suggests good but not quite perfect.

Anyway, electrostatics have their limitations too, which may be more bothersome to many who like slam.  These can be overcome in a reference system, like the one I'm still working on, with additional time aligned drivers for deep base, super highs, and so on.  Without those complicated fixes, whether one prefers electrostatics or single cone Audio Nirvana is a matter of choice, neither is perfect really.

So the Audio Nirvana approach makes a truly excellent Value system (by Audiophile standards of Value…non-audiophiles would think you were already spending enough for Reference).  But it may not offer as much potential for improvement as a system based on electrostatic panels plus additional drivers like mine.  I imagine that Electrostat loving friends of mine might be horrified at the idea their speakers are lacking anything, but I think it's a mainstream perception anyway that you get maximum slam with a dynamic loudspeaker system.  People like me try to get the best of both, by mixing both.

It's kind of a miracle that they've made single dynamic drivers these good now.  Most people would not think it possible.  I didn't.