Friday, December 26, 2014

Why not to bother with listening tests

Peter Aczel suggests not bothering with listening tests on well performing electronic equipment, and now even speakers (not to mention special cables and tweaks, which he says have no effect).  Measurements and basic design tell you pretty much all you need to know.  There is no scientific evidence (such as with double blind tests and proper statistics) that shows otherwise (wrt electronics anyway--Peter goes by his own measuring/listening experience wrt speakers).  Box speakers with lumpy response will sound like box speakers with lumpy response.  (Last I read, Peter had gone on/back to dipolar speakers…which have some key advantages such as not exciting room modes and spurious room response generally.)

Peter can be a bit over the top.  But against the matra, "listen for yourself," I think he is almost entirely correct.  There is most often little or even negative value in "listening for yourself."  And I can say different things than he does.

Listening to speakers remains a good thing to do, actually, with an open mind not prejudiced by previous listening experience, and with great disregard for the consistency of your listening results.  But not necessarily so much to each and every little modification to speakers, electronics, or anything else.   Unless there are likely to be large differences in measurable amplitude, such as 1dB or greater, listening tests are likely to be wrong.  And even if there are large differences, they may be misinterpreted. Listening tests to things that have no explanation acceptable to audio objectivists (like Peter Aczel) shouldn't even be bothered with for most people--except people who thing strongly otherwise, and they should concentrate on getting definitive double blind test results to prove their position rather than engaging in endless polemics.

The principle reason why you should not even bother with listening tests (except, I'd concede, on speakers and phonograph systems--if you bother with phonograph systems) is that the outcomes of listening tests are untrustworthy.  There is huge variation in auditory experience ever under identical and controlled circumstances.  This variation likely explains all the differences allegedly heard in comparisons which have no explanation Peter Aczel and audio objectivists at Hydrogen Audio would accept (and they're pretty tough).  And probably a lot more, most untrained listeners (and I'd consider myself barely trained if at all) don't necessarily even reliably hear differences at the scientific thresholds of human ability, such as level or frequency response differences of 0.1dB.

As a first order of approximation, if the very serious listeners at Hydrogen Audio don't believe there is a difference they can hear, you and I, less serious, should probably not even waste our time.  We'd need to spend months training our ears to reliably hear and identify small differences as serious DBT aficionados or academic audio engineers (who are more interested in truth than having a competitive angle).

To rise above the untrustworthiness, if you believe otherwise, prove it to yourself and others with a double blind test.  I tested myself in 1983, and since then I have become far less serious about audio tweakery (although, as you can see reading these pages, I have hardly given up on tweako-explorations of the kinds that make sense to me, generally involving physically identifiable and in-principle measurable factors, just beyond the normally accepted limits).

I find fancy cables and capacitors and digital converters fun to think about.  I doubt I could prove scientifically that I hear differences, even after a long struggle, though perhaps could with actual lifetime training.

But if I find them fun to think about, why not have them in play anyway?  Or why should I deprive myself the fun of buying and deploying fancy gizmos regardless of my doubt over their efficacy?  Why should people with foggy thinking on this subject have all the fun?  I temper my lust for gleaming machinery with the serious knowledge that it's all really bs anyway, meaning basically I can spend only what I want and on the things I want--not the things other golden ears have come to believe in and insist I must have.  My more critical thinking only advances my freedom to do anything I want.

Now given very high unreliability, frequently hearing differences that can't actually be heard, the psychological effect of performing listening tests routinely is to develop audio superstitions.  Audio superstitions are acceptable to a certain degree, just like all psychological disorders, we all have a little of them, and perhaps life would be boring if we didn't.  But for some people, superstitions become very costly in a variety of ways, not necessarily even including money.  Superstitions can become costly in behaviors, including not-listening-to-music (the well known audiophilia nervosa), but into many other obsessive and anti-social behaviors.  People may begin summarily dismissing their closest friends or associates because of petty disagreements based ultimately on dueling superstitions.  Obsession may lead people to abandon their friends and gainful employment, in order, usually nowadays, to conduct endless internet flame warfare.  And there can be various kinds of paranoia, grandiose delusions, and insomnia.

It's not as though, necessarily, doing listening tests regarding audio minutae will lead immediately to psychosis.  In fact, you could do all the listening tests you like, either if you proceed scientifically with dbt, or, universally, simply fail to take yourself and your perceptions at any given time very seriously.

Put then if you don't take your momentary perceptions very seriously, why would you even bother with listening tests?  Listening tests, especially serious listening tests, take serious time and effort and generally speaking are a big boring head scratcher.

A standard methodology for non-listening-test audiophiles like me is basically buy stuff and hook it up.  If it sounds bad, replace it with something else.  If something new reads interesting to my collection of audio knowledge, superstitions and lusts, repeat.

Lack of believing in listening tests doesn't mean I can't buy and play with cool stuff, just based on what I know about them through other means than personal listening tests.  It doesn't mean I need only restrict Myself to that which I know as provably audible.  I need only be conservative and reasonable in my expectations and findings to avoid madness, and stay merely eccentric.

Monday, October 20, 2014

How Low Bass? As Low As You Can Afford

Here's a discussion on how low bass should go.  Keith Yates (who did great investigations of deep bass and reviews of subwoofers a few years back) recommends 8 Hz.  Black Hawk Down has a 7 Hz pulse which people say makes your fight-or-flight instinct kick in.  Pretty much everyone in this discussion (except R.D. Clark after his heart surgery) loved deep bass, and the deeper the better.  Many suggest you should go as low as you can afford, it improves everything, etc.

I have no trouble reproducing 16Hz in my main living room system, or my bedroom system.  But the problem in the living room system is room modes and rattling.  There are in fact resonances around and below 20 Hz, perhaps those are volume modes (not the dimensional modes usually discussed), so when I equalize the bass for full output I get more annoying rattling.  And when I flatten out the worst mode around 45 Hz, the bass tends to sound anemic even if it goes deep, since most actually recorded bass is above 32 Hz.  Also, a kind of congested sound arises from boosting the bass to acoustic flatness below 32 Hz.




Thursday, October 16, 2014

Multiple Subwoofers

Here is an interesting discussion of multiple subs.    To maximally reduce the effect of room modes, they should be widely spaced.  Diminishing returns sets in after 3 widely spaced subs.  Above the Schroeder frequency modes are packed close enough together to become unimportant.

Friday, September 12, 2014

Toslink Upside Down

Nobody seems to talk about this, so I shall fearlessly proceed.

Toslink connectors are supposed to go in only one way.  However, in my experience, most Toslink connectors can get plugged in upside down.  Generally you have to apply unusual force to do this, and the instructions for Toslink connectors always say that you should not "force" the connectors.  So it is clearly wrong.

But sometimes very tempting.  When Toslink connectors are forced in, upside down and all the way in, they become very rigid.  Plugged in the usual way, Toslink connections can sometimes be very loose, sufficiently loose that with some equipment the signal can be unreliable or intermittent.  Though I can't recommend that others do this, I have chosen to plug in connectors like that upside down.  Currently this means the two Toslink connections to the first M-Audio CO2 in my kitchen.  If I plug those connections in "the right way" they become intermittent if I wiggle the cables.  That is not good for stuff that can get moved around on top of the kitchen table.

But I worry and wonder if plugging Toslink connections upside down damages either the cables or the equipment being plugged in.  It could in fact be that the reason my first CO2 requires cables to be plugged in upside down is that I did so sometime in the past, wearing out a certain part of the connection socket so that ever since it has been intermittent for cables plugged in the right way because it doesn't hold tight enough anymore.  Alternatively it could be that I damaged the connectors on the cables themselves in this way.

So I can't recommend that others plug in Toslink upside down, but for now, I am doing that for certain connections.

I am very glad that my new Schiit Modi DAC (OptiModi) has a Toslink connection that is not unreliable if the connector is wiggled, when plugged in the correct way which seems slightly loose.  I suspect that is the way they should be by design.  The connection should not have to be rigid to have a reliable connection.

But that doesn't seem to be the case with my CO2, and it also was not the case for an Inday TLDA22 Toslink splitter I bought, which introduced me to the idea of plugging in Toslink upside down, for it would barely work any other way.  It's possible that once I started plugging the cables into the TLDA22 upside down, it damaged the cable connectors so that now they won't go into the CO2 well enough the correct way.

I do have some Toslink cables with metal connectors and I think those would not be so easily damaged, though it might cause damage to the equipment.




Thursday, September 11, 2014

Good Schiit! (OptiModi)

I'm very pleased with my $99 Schiit Modi with Optical interface (OptiModi).  It looks, works, and sounds great, and is an incredible bargain!!!  Also made in USA, in a nice little metal box, audio jewelry on the cheap!!!

Unlike other devices I've had, the Toslink suffers no dropouts when I wiggle the connector.  It doesn't hold the Toslink connector as tight as I like, but no matter if it's completely insensitive to connector movement, as all Toslinks should be, but some aren't (my next post discusses Toslink connectors).

Loaded with top silicon, such as one of the latest converters from AKM, it may be the best sigma delta DAC out of a few dozen in the house.  I won't be testing it against my R2R 1704 DAC for the living room panels.  I am slightly afraid I might think this little job was better…

The night I first plugged the Modi into my Kitchen system, I was rocking out to the good sounds it made.

I plugged it into my Pioneer DVR-LX70, as it has always been intended to provide the analog audio that machine requires.  My living room receiver accepts digital inputs, so it seemed natural to provide it an optical input.  And so I have long used some kind of optical splitter on the Mac output to produce one Toslink for the receiver, and one Toslink for a DAC which produces the analog required for a DVR.  The DAC has always been the Emu 0404 USB I repurposed as a DAC for this use, temporarily, in 2011 or so.  I really intend the 0404 for use in making electronic and acoustical measurements.  It's been a pain in the neck because after each power outage, the 0404 has to be reset (since on power up, it goes to "OFF" on the main selector.  And then I can't remember if I need to do to get it working again.  (I strongly dislike DACs that forget their last state on power cycling.  My Aragon DAC was like that and I hated it.  Then it died.)

So the purchase of the Modi was really about streamlining and making my Kitchen system more robust, rather than "getting better sound."  I had every reason to believe the 0404 was top shelf, didn't really need replacement.  And further, that routing an analog converted signal through the LX70 and thence to my receiver in analog form would not, could not, be as good as sending the receiver the original digital.

But it was sounding so good as I was listening to Abby Road mixes, I was thinking my thinking had been wrong.  If the Yamaha 5790 is fed analog in direct mode, it never goes through the Yamaha's ADC and DAC.  It goes through in pure analog, "Pure Direct", which is made possible by having a parallel volume control that controls the DSP when doing digital processing, and an digitally controlled amplifier (an analog circuit that does not sample or quantize the input, but is controlled by a digital signal).  I tested and determined that the receiver does actually operate this way when I received it.  In "Pure Direct" you see no digital artifacts at the output if you provide an analog signal without digital artifacts (typically a waveform generator).

So perhaps the better sound (or at least heightened experience) resulted from the superiority of the Schiit Modi over the DAC built in to the Yamaha.  Of course Hydrogen Audio objectivists would say this is impossible, modern DACS and ADC's are completely transparent, and moreso than much analog circuitry.

Anyway, on Thursday I compared the two DAC's more directly bypassing the LX70 (which may be boosting the bass a bit--Legato Link???) by plugging the Modi straight in to the Yamaha.  While I initially thought there was a difference, as I flipped back and forth after the first comparison in critical listening I found there wasn't really any way to distinguish the two.  Finally I got very bored at the possibility of finding a difference, they were just too close.

So there you go.  At first I though there was a huge difference, the Modi was a "revelation", opening up music better than ever before, perhaps better even than my cherished 1704 dacs.

Later I find no reliable difference between the Modi and my receiver, which would exactly be an audio objectivist belief, all DACs sound the same.

It's easiest to have a strong belief if after doing A and B you then quit testing.  Going back to A again confuses things.  I have always found this.  But going back to A perhaps also reveals that the difference  experienced between A and B was not due to the stimulus being detectably different, but rather the state and expectations of the listener.

The Schiit Modi is perfect for what I bought it for, and perhaps more.





Left A'Diva Speaker Hung

Finally got around to installing the left speaker in 2nd bedroom.  Not hooked up yet because I plan to use flat wire that sticks to wall.  The wire I have is labeled Aurum but looks identical to Ghost Wire sold by Sewell.  Here are the instructions for running Ghost Wire.  My plan is not to use the big terminal blocks sold by Sewell on the grounds that they are big and ugly.  Instead, I plan to fold over just a bit of wire at the end (that way I can get behind the sticky part on top surface) and solder small 16 or 18 gauge wires to it.  At least up by the speaker.

Tuesday, September 9, 2014

What to buy next

Back in June when I attended THE Show in Newport, I decided that after my second vacation in July I'd get myself a DSPeaker since bass boom and/or lacking bass response is the single biggest problem in my living room system.  But then before my July vacation, I decided to buy a DVD-9000.  Issues related to the DVD-9000 continued through about mid August.  So then the window opened up again.  I'm home for the foreseeable future (no planned vacations for the next 10 months or so) and I can buy stuff now.  But by this time, I was no longer committed to buying the DSPeaker Dual Core, having found issues with it and other approaches which might be better (and, unusually, cost less).  What I really need to do is start making full system measurements using REW or something better.  In August, I picked up a nice $100 calibrated microphone with bass calibrated to 5 Hz.  I could get started on this at any time now.  I also did a system time alignment and tuning using Tact for measurements (but not correction).  But an independent fine resolution measurement with REW would be better.  I just have to get around to doing it.  Note: meanwhile I have been doing many other things and making great strides forward if not in the direction I planned in June.

So I'm now putting off major purchases until I do that.  Especially with regards to EQ or acoustical products.  I need to establish baseline measurements as well as get comfortable making those measurements.

After doing the measurements with REW, I may simply choose to adjust and add to the Parametric EQ's (PEQs) in my Behringer DCX crossover or DEQ equalizer.  Another alternative would be to get an OpenDSP product to implement filters designed using REW.  Existing commercial products Dirac and Accourate don't look like what I want--I might prefer the algorithms used by DSPeaker to those.  But in either case, I want to do the measurements first.  Then, audio things I could buy would include:

DSPeaker Dual Core 2.0  (after all)
OpenDSP-DI (an alternative that works with REW)
Bag End E-Trap (an alternative way of taming modes, and it might be best for problem spot I have)
RealTraps, GTK, etc: Mondotraps, Minitraps, etc

Those are specifically things to deal with bass boom and impact.  But meanwhile I have other audio projects going.  Just last winter I started the Turntable Project, which has been on hold since about April.  Things for that project include:

new 12" tonearm and arm cable for Lenco table
new extra phono cartridges: Dynavector 17D3 (I could use up to 2 more!)
second moving coil amp for living room
repair Technics EPA-100 tonearm
refurb Technics motor
dustcover, dustcover
Get LP12 fixed (requires new cartridge)
Get Sony PS-X800 fixed
New turntables ???

And then, that's hardly the beginning of audio projects I've been thinking about:

New R2R Dacs for super tweeters, subs, and master bedroom system (4 new Dacs in all)
New Behringer DEQ's to replace DCX's so I can use external DAC's
OR, modified Behringer DCX's (and DEQ's)

I'm sure there's more, but that's about all that's coming to mind now.  Meanwhile, I also hope to get some big home improvement projects done, starting before January, and on for the next 4 or so years I hope to get these things done:

New master bathtub and tile (high priority!)
Other bathroom remodeling (lower priority)
New driveway extension
New patio and patio cover
Garage/Gym redesign
Other kitchen and bath upgrades
Solar system

And, in addition to that, there are some other important expected purchases in next couple years or so:

New couches for living room back wall
New kitchen chair for 2nd person
New adjustable bed for Queen's Room

Finally, in about 2 more years, I'm going to need a new CAR, for which the least expensive model I'm considering is Nissan Leaf for about $35,000.

With all this on the buffet table, I'd better not scoop up too much audio bricks.  A sensible person might say I've already got enough audio stuff…

Analogmetric

Saturday, September 6, 2014

Toslink and SPDIF raw signals compared

At the September 2014 XCSSA meeting I brought a box of SPDIF Coax to Toslink converters, analog to digital converters (to generate digital audio signals), and an oscilloscope, all to see how the raw Coax/Toslink signals look in various conversion configurations, including ultimately 3x conversion.  Direct SPDIF coax is better than using Toslink conversion, as I expected.  But the differences are quite small, and it looks like many levels of conversion would likely be OK in a digital audio system.  I have been using 2x conversion (actually, 4 conversion elements) in my latest digital audio system at 96kHz, and even with other long cables, it has worked fine, and has sounded great.  Based on what I saw at the meeting, that is not surprising, though I also demonstrated that a new method of hooking up converters will likely work better than what I have been doing in the past--but I did not have enough equipment to test that directly.  I'll discuss what I mean by conversion and how it should be counted as I go along.  I also showed that a bandwidth measurement for digital audio transmission should likely go to 20-100 mHz.  The 2mHz oscillator I brought was simply too low frequency to give any digital audio transmission systems a meaningful test.  The actual 96kHz digital audio signal seemed to use pulses equivalent to a 6mHz square wave, from our Oscilloscope estimation (which I haven't verified).  But the risetimes suggested transmission bandwidths in the 35-100mHz range, even with multiple levels of optical conversion.

First off, I connected a Analog to Digital converter to a generic brand 12 foot video cable, and connected that to the 100 Mhz oscilloscope having a special 75 ohm terminator.  (I am so lucky to have one of those!)  The square pulses of the digital signal are almost perfectly clear and square.


 I replaced the 12 foot generic cable with a 6 foot Monster Cable Video 3  75 ohm cable.  This specific model was recommended in a Widescreen Review technical test as having good performance.  (The Monster Video 2 was only so-so.)  It seems to use a solid center conductor, and uses slotted center pin metal plug.  It had performance essentially indistinguishable from the 12 foot generic cable.


After doing that test, I cleaned up the picture slightly by reducing intensity and adjusting focus.  If I were trying to fool you, I'd delete the second picture and just say this is what the Video 3 cable response looked like.  And that's true.  But it wouldn't be honestly explaining what caused the difference.



Now the picture is so clean we're seeing high frequency ringing which may be occurring in the SPDIF generating circuitry.

As shown, I think that coax cables 12 feet or less in length actually make very little difference to SPDIF connections.  So I won't be paying attention to which coax cables I'm using in these tests, and in any case they were all Belden 75 ohm video cables of one kind or another, terminated with Canare 75 ohm RCA connectors, in lengths of 1 foot to 7 feet.  I also didn't pay any attention to the optical cables I was using since they were all short, 6 feet or less.  Long Toslink cables might have some ill effects, but I did not bring any long Toslink cables with me.  I have used 30 foot Toslink cables with good results, but 30 feet might be getting to the point where some degradation of the signal sets in.  But I won't know that from these tests.  What I believe makes the most difference are the Coax to Optical and Optical to Coax conversions.  And possibly if you used really bad cables or connections, which I didn't do.


But the third scope picture only looks so much cleaner because of scope adjustments.  BTW, adjustments in future pictures will vary, because I need to readjust intensity a lot when going to very high speed pictures like the one below.  Notice above that the rise time appears to be about one tick mark in a grid having 5 tick marks per 0.05 uS.  So one tick mark would be about 0.01uS rise time.  Bandwidth is calculated as 0.35/rt.  Funny that linked page does exactly the calculation I need, and the result is bandwidth of 35mHz.  Much of the bandwidth limitation is likely in the Analog to Digital converter, which doesn't need a 10Ghz buffer to drive SPDIF at 6mHz.  Some is also in the scope and cable.  The scope is only a 100 mHz scope and likely sufficiently out of calibration to be somewhat less than that.  To verify the 0.01uS rise time I pressed the 10x horizontal button.


Well, actually it's better than 0.01uS.  Each major division here is 0.005uS, and it seems that the rise time is about that.  So now it looks like 70mHz bandwidth.  We're possibly getting close to the limits of the scope here, so the actual coax bandwidth might be still higher.  Notice that the top squiggles seem to cycle in about 0.01uS, which mean they represent a frequency of 100mHz.

Now I do one set of optical conversion with two conversion elements.  I take coax from the ADC and run it through a M-Audio CO2 converter to convert the coax to Toslink.  I run a 1 meter Toslink cable to a second M-Audio CO2 to confer to Toslink back to coax, and then to the scope.



You have to look carefully to see any differences in the scope picture.  But the 100mHz squiggles down the whole top of the pulse are gone and there's a bit of rounding at the leading edge of the signal pulses.  Zooming in, we can see the rise time is now about 0.009uS, so the bandwidth has fallen from 70mHz to 39mHz.  One would doubt this would be a problem for any digital audio transmission at 96kHz.  (Heck, even very ugly looking digital signals work, and this looks nearly pristine.)



Next I added a second set of optical conversion, by inserting an active optical splitter in between the optical sender in one CO2 and the optical receiver in another CO2.  I've had good luck with this active optical splitter, an Inday TLDA1.  Don't bother with passive Toslink splitters, they usually don't work.



 After inserting the splitter this I had to readjust stability controls in the scope, and the resulting stable pulse is a different one than before, one surrounded by blank pulses and seems narrower--ignore that difference.  I also might have changed the time base a bit to get stability.  Otherwise, t looks about the same, though you can see some roughness in the top and bottom curves of the pulse, which looks like some kind of added vhf ringing.




As much as I've always thought the Inday Toslink splitter was a surplative product, here it appears like it adds some artifacts compared to using a pair of CO2 converters.  Though by necessity I can only test the Inday in combination with a pair of CO2's, so I can't really isolate their effects.  But still, I now believe CO2 is better because of cleaner signal and lack of stability issues on the scope.  (BTW, the M-Audio CO2 is no longer being made.  Isn't that the way things go.)

I tested a different model Inday Toslink splitter, TLDA22, one which gives two optical and one coax output from a single Toslink, a feature I thought was cool, but I had many problems with this unit (and for that reason I have gone to great lengths to duplicate the functionality of the TLDA22 using multiple CO2 devices).  Running just as the previous Inday splitter, using Toslink input and Toslink output, and forcing the Toslink cables in upside down as they are too loose the correct way, I got an essentially identical trace from the 96kHz digital.  I had previously believed this TLDA22 to have broken 96khz capability, but here it seems to work as well as needed, except perhaps for having similar stability issues as the TLDA1 on the scope.


Now I couldn't run all three CO2's I had brought at the same time because I was missing the 9v DC adapter for it (did I not read the eBay ad carefully?).  We couldn't find a suitable 9v DC adapter in the large box of AC adapters at 10bitWorks because the few we found didn't have room for the large center pin in the connector.  9V DC adapters are relatively rare, actually.

But what I could do was use a different SPDIF generator that has both SPDIF and Toslink outputs.  That was an Emu 0404 USB, running in standalone mode.  Selecting the analog inputs and connecting to the Coax output, it produces pulses with a slightly curved top, suggesting some HF damped resonance around 10mHz, but still a clean looking wave (and it was very stable, note this was a lower sampling rate, probably 44.1kHz, since I can't vary that on an 0404 in standalone mode).



Now I remembered to test one of the CO2's in coax-to-coax mode.  The CO2 has a selector switch which can select Bidirectional (the usual choice), Coax-to-Toslink, and Toslink-to-Coax.  In Coax-to-Toslink mode the Coax input is converted to Toslink, and, simultaneously, the Coax output operates as a coax pass-through (it's an active splitter, actually).  Inserting the a single Coax-to-Coax into the 0404 output chain only made the pulse look better--with a flat top.  The rise time (and even ringing, in a later test) appeared identical.  Thus the Coax-to-Coax looks better and cleaner than any kind of optical conversion (but I should have chosen the better test signal, arrg!, because on this uglier test signal you can't tell as much):



Next I did something almost opposite.  Starting with the Toslink output of the 0404, I ran that to a CO2 in Toslink-to-Coax mode.  But I connected from the Toslink output of that CO2 (remember both Toslink and Coax outputs are active in Toslink-to-Coax mode) to another CO2.  The second CO2 simply converted Toslink back to coax for the scope.  Now this chain actually involves 2 layers of conversion or more precisely 4 conversion elements and one buffer.

Layer 1:
internal signal to Toslink output inside 0404
Toslink to internal signal inside CO2 #1

Layer 2:
internal signal inside CO2 #1 to Toslink output
Toslink to internal signal inside CO2 #2

Buffer
internal signal inside CO2 #2 to Coax output


About the same, though it does look like the rise time has fallen about as much as it did in the first test using 4 optical conversion elements, though the digital audio generators are different and the tests are not exactly comparable.  Still, it looks perfectly fine.

By this time I was getting confused and decided to do three tests in a row.  Starting with direct from coax from the 0404 to the scope, then taking the optical output of the 0404 through a single CO2 to coax (one layer of conversion, including one element inside the 0404 itself), then the two layers of conversion (same as previous experiment above).  You can see some progressive deterioration but not much.  The CO2 converters are so clean I don't believe I messed with the scope stability controls at all during these three presentations, though perhaps a tad on the last one.






I tested running the Toslink output of the 0404 into the Inday TLDA1 splitter, then through a CO2 for conversion back to coax.  This is two layers of optical conversion again, same as in the last picture above, but with one of the layers occurring within the Inday TLDA1.  The result looks to have slightly higher rise time, perhaps, but slightly less clean than the double conversion plus buffer above using two CO2's.




Now for the most conversion I was able to do.  Triple layers with one buffer:

Layer 1:
internal signal in 0404 to Toslink output
Toslink to internal signal in Inday TDA1

Layer 2:
internal signal in Inday TDA1 to Toslink output
Toslink to internal signal in CO2 #1

Layer 3
internal signal in CO2 #1 to Toslink output
Toslink to internal signal in CO2 #2

Buffer
internal signal in CO2 #2 to coax output



Despite all the conversion, it still looks pristine, bandwidth at least 35mHz, etc.  The Inday splitter adds a tiny bit of ringing, hardly noticeable, but also seems to make the rise time even quicker than without it.

Using the other Inday splitter with coax output, it was hard to get good triggering with 2 layers of conversion.  It occurs to me that scope triggering is much like the triggering inside a digital receiver in that it focuses on the transition from very small negative voltage to very small positive voltage.  It doesn't look at the scope trace like we do.  What I could see beyond the lousy triggering was that the rise time seemed just fine, perhaps even better than the CO2's.  But the freedom from ringing makes me prefer the CO2 devices to the Inday devices.

My ultimate hookup for Mac to Living Room Stereo will use two CO2's.  The optical output from Mac computer will get converted to coax and optical (Optical->Coax) by CO2 #1.  The coax output of CO2#1 will connect to the second CO2, in Coax->Optical mode.  Then I will have two optical outputs, one from CO2 #1 and the other from CO2 #2, and one free Coax output, from CO2 #2.  The optical outputs go to devices in the kitchen and the Coax output goes through installed wiring to living room.



I had been concerned that the second Toslink outputs would see three layers of conversion, but now I see that is not true.  The Toslink output from CO2 #2 sees these 2 layers of optical conversion, plus some coax buffering (which I have determined to have little ill effect):

Conversion Layer 1
Mac internal signal to Toslink output
CO2 #1 Toslink input to internal signal

Buffering Layer 1
internal signal to Coax output
Coax input for CO2 #2 to internal signal

Conversion Layer 2
internal signal to CO2 #2 Toslink output
Toslink to internal signal in receiving device

So this is only two layers of CO2 conversion, which I tested in many ways above.  It is not as bad as three layers of optical conversion, which also looked good in testing.

Late on Saturday night I hooked up the two CO2 converters as described above, but the second optical output is not currently connected to anything yet because I will use it to install a Schiit Modi DAC.  The other outputs are working perfectly and sounding great, especially the living room connection which now sees only one layer of optical conversion while previously it saw two.

Friday, September 5, 2014

I like Science, but this blog is for Fun

I pretty much agree with Peter Aczel on what is audible (I say, likely audible) or not, and the people of Hydrogen Audio and so on.  I greatly respect all of them for digging for the provable truth, scientific and so on.  DBT is definitely the way to do things scientifically and work toward a believable collection of ideas.

But I'm running my audio hobby for fun, and DBT is just plain hard and boring work.  I prefer to do things I like to do, which generally doesn't include testing of any kind (and I figure if you're not going to do DBT you might as well not do any testing at all) so I usually don't bother doing any, I just plug the new stuff in and go, unless it's really bad.

So this is not about science, truth, and what is provably true.  I'm only moderately interested in getting by with the least expensive amplification I can and so on.  Actually I have a far less expensive amplifier than Peter Aczel, and probably one he would have recommended (though not as much as his).

This blog is not about "science in the service of art," much as I respect that.  Recorded music is nice, mind expanding, and so on, but it's only one thing I do.  I also spend a lot of time messing with things which may or may not be all that important (in fact, they probably aren't) but I have fun doing so.  I like cool stuff, and I like overbuilt audio equipment, and technically interesting audio concepts.  I have fun thinking about such things.  I don't claim the mantel of science.  I claim the mantel of play.

One of the best commenters on Hydrogen Audio, the famous inventor of abx testing, has admitted if all you are doing is fooling around--which is exactly what I'm doing in full awareness of that--go ahead, you are free to amuse yourself however.  What he finds fault with is falsely claiming your testing is valid, scientific, etc., when it doesn't pass DBT basics.  I agree with that completely (though I still read TAS and Stereophile for amusement).  And I fit his OK category of someone who knows and will readily admit that what he is doing is not scientific, etc.

BTW, I have done blind testing on myself, and formal DBT's on others, on audiophile theories.  Not one of these tests had a positive result.  This was somewhat mind blowing for me, and I think it should have been for others.  I know testing on anything having peculiar audiophile interest, like some of the things reported on here, would be very difficult.

I don't believe in most audiophile testing.  Sighted testing is likely worthless or worse.  I believe in scientific tests such as those at Hydrogen Audio, but those aren't very interesting for many kinds of fooling around.

Since sighted testing is likely worthless or worse, I don't bother taking it seriously.  I indeed make pivotal judgements sometimes--quite often--listening to background music--even from another room.

This is all for fun for me.  I do hope something cool eventually turns up, but meanwhile, I know I can't make any special claims.  Except many people have said my sound is the best, and I myself think it is up with the best I saw at Newport 2014, such as the grand MBL system (with many qualifications of course, my room is far smaller).

But there are problems.  My bass EQ is ad hoc and incomplete, better room treatment is needed (I've been thinking Bag End eTrap, I also like RealTraps, funny the many camps in room tuning.  Even Ethan Winer says some EQ is necessary.  He puts down the eTrap but not seriously IMO.  It's a interesting question how much aborption would have the same effect as an eTrap in the very corner, at the most crucial frequency.  I have a corner in mind, the first hallway corner which is just 8 feet from the subwoofer.  I had an outlet put there last year, eTrap was on my mind.  There is no space for a serious corner trap.  I could put a pair of mini traps vertically.  But more than that would cut down the hallway space too much.  My belief is that the eTrap "cancels" more of a serious mode, when placed in the required corner, than a pair of mini traps on the wall vertically, at 45 Hz.  The mini traps would have the advantage of greater broad bass band absorption, but less peak absorption at the critical frequency.  No where around my room in the lower half do I have room for a serious bass trap.  Only around the ceiling, perhaps, an installation challenge.

Nowhere on the web I have seen a serious, "scientific" comparison on room adjusting methods, including EQ (there are now many comprehensive and semi-comprehensive EQ systems, not just the one or two points Ethan Winer suggests), room damping, and active devices.  The latter two could be a very direct scientific comparison.  How much corner treatment works of all kinds could be directly compared, measured, DBT's.

I might at least end up doing the measurements, if I acquire both kinds of things, which I think I will.




Updates

Thought: It is sometimes assumed that because some part of an audio system is not up to standards, the rest need not be.  But actually a weakness in one place is multiplied by weaknesses in others, so the quality of the remainder is more critical when the preceding is poor in some way.

Played two albums on the bedroom MT-30.  I liked the sound, in fact thinking it better than when I last heard the albums, probably on the Sony PS-X800: Friendly Neighborhood Big Band, and then King James Version (D2D boxed version).  After hearing the first, I was thinking, OK, maybe 45 RPM isn't so bad, but the second album proved that 33 1/3 is sounding good also.  Maybe the old caps are reforming a bit.  I still think this table should be recapped and motor tuned.  The actual motor is apparently unserviceable but OK.  But the electronics is now 34 years old!  Anything that old should be fully recapped and restored.  Why better sound?  I think mainly the vinyl plank flooring with Quiet Comfort Premium underlayment sounding better than old carpet, and other improvements.  Not so much the table perhaps.  I'm still thinking that the Sony was better, when working.

Also thinking of the irony, I became an audiophile around the time the Sheffield Labs D2D recordings were being released.  I was convinced that reel tape was a horrible bottleneck.  Now I think of master tape as being beyond all, at 15ips even.  But anyway, D2D was an interesting stunt, and actually any reduction in signal path is an improvement, and the D2D involved lots of skipped intermediate steps and enforced simple recording and processing.  So the records are great sounding.  Very hot too.  Given that you are going to listen to LP, D2D gives the best possible sound, if possible, as usually not.  If it were D2D vs 2nd generation 15ips, I think the tape would win.

When at first I replaced the cheap stranded video cable I was using for the digital audio line from kitchen table to patch panel with a nice new custom Belden 1505F with Canare connectors from Blue Jeans Cable, the first thing that happened was that 88.2k wasn't going through.  I switched back to old cable and same problem.  Then I remembered the trick I had done at first, always start with 44.1 then switch to 88.2.  So I'm using the new cable now, as it worked at least as well as the old one.

But on Tuesday night I replaced the living room part of the line: the series of cables from patch panel to Tact preamp.  First, the F-to-RCA adapter, then a 6 foot Monster Video 3 (an OK cable), then a barrel connector (likely not a good idea), then a Radio Shack premium video cable (OK but not as good as the Video 3).  That whole slew got replaced with one cable, custom from Blue Jeans Cable, a Belden 1694a with F connector on one end and Canare RCA on the other.

Well when I connected that cable it started right up in 88.2khz, no problem.

But wait, isn't the main issue still the double 2-way optical conversions in the signal path?




Friday, August 29, 2014

John Siau on DSD

This is a great read for DSD non-believers like me, and I think I've linked it before also.

John Siau does not say that DSD is a bad distribution medium.  Like people I've read at Hydrogen Audio, Dr. Siau believes DSD to be roughly equivalent to PCM.  But it creates many issues for mixing and mastering, as well as end user DSP for DSD purists.  Siau says conversion of DSD into PCM is fine, it's the conversion of PCM into DSD that raises issues.


Thursday, August 28, 2014

New Sonos in 2nd bedroom sounding great

Gallo speaker mounted in right upper corner of bedroom 
In the past week I've fixed two fundamental problems with Sonos in the 2nd bedroom.  First, I made it possible to get the ethernet connection from the Kitchen (which has a Dlink fast managed switch that now directly connects to all Sonos boxes in my house) instead of the Computer Room (where the old internet router creates a bottleneck for the whole network), by changing the HDMI over Cat6 video extender to a better model which only requires one Cat6 cable, leaving one of the Cat6a STP lines from Kitchen to 2nd bedroom available.  Then, I figured out how to fix the persistent ground loop without ground lifting the Parasound Zamp V.3 amplifier--by attaching the Sonos box with a short length of unshielded ethernet cable.  (Currently using an ethernet coupler at the end of the existing 10' Cat6a STP which attaches a 5 ft Cat5e UTP, the best short UTP I could find on hand; I have purchased online a 10' Cat6a UTP which will ultimately replace the 10' STP.)

On Tuesday I went even further and figured out how to get mono out of the Sonos box with two Harrison Labs attenuators and a Y adapter.  I also neatened up the cabling somewhat (though you might not see that from the picture) and tucked the bare metal shielded ethernet coupler (possibly having some inductive leakage, though I couldn't feel any) in between the gear so it would not be touched by dog or cat under the desk.

Zamp, Sonos, and outlet strip installed below small desk

It makes a huge difference to get the full L+R output rather than just Right Channel even though there is still only one speaker.  I've started working on mounting the Left speaker but it may not be fully hooked up for awhile because I'm planning to use "invisible" paintable flat cable.



I was enjoying listening to KPAC over the Sonos in this room, in true mono L+R, while simultaneously playing line inputs through all other Sonos boxes in the house.  Using the line-input feature in Sonos is the hardest test of Sonos networking, and in my experience, the most valuable usage of Sonos.  In one test, I was even playing the input from the master bedroom on the 2nd living room box, even though that box isn't hooked up to anything.  With all 5 boxes in line-input play on other boxes, I might have heard some glitching while I was brushing my teeth (so I couldn't be sure it wasn't the station having trouble at 5am).  When I rolled back to the 4 boxes I actually play from, there hasn't been any glitching in 3 days.  So the new network design using a fast switch in the Kitchen is now a proven success.

(When first setting up the 5 line-input test, the living room box hung and needed rebooting.  That box has needed a lot of rebooting over the last few years and possibly should be replaced for better health of the entire system.  Since the last reboot, however, everything has been running fine and glitch free for days using 4 boxes for playback of line inputs on other boxes--except that the living room box has only been used to play it's own line input.)

Tuesday, August 26, 2014

ZSYS

Dirac vs REW vs DSPeaker

This is quite a long blog about Dirac at WBF and it was all worth reading.

DSPeaker like REW appears to be minimum phase and IIR based.  Dirac is mixed phase and so can correct some additional time delay problems.

I just can't use full range correction with electrostatic speakers because they are very complex in the high frequencies.  Tiny changes in position make huge differences in response.


Sonos Hum caused by STP cables

I'd started to think it was inevitable that there would be hum in the 2nd bedroom Sonos unless I ground lifted the amplifier.  The previous version of the Parasound Zamp V.3 did have a ground lift switch, but mine does not.  It seemed possible that it had been shipped with a 2 wire power cord, but I have been unable to get that confirmed.  I tossed the original power cord into my spare box without checking until later, and when checking later the cord that most likely came with Zamp has only two connectors on the plug (but three slots in the IEC female portion that connects to chassis).

Ground lifting the amplifier might not meet electrical code UNLESS the amplifier was supplied with the ground lifted cord.  And even then, a paranoid electrician might say you are better off having the actual ground connected, which the 3 connector IEC on the chassis permits.  Some equipment is deliberately designed with 2 wire IEC connectors which can't be grounded no matter what kind of cord you attach.  But not in this case, grounding is clearly an option, and an option which would be better used if possible, any good electrician would say.  It protects both against breakdown of the insulation in the amplifier transformer, and current which might be carried inadvertently by the network cables (say, if they had an insulation failure in the attic).

An extra line of defense in my case is that the circuit to which the amplifier is attached has an upstream GFCI outlet which provides GFCI to this outlet.  I tested it with a GFCI tester to be sure.

However, despite all my precautions, my friend who is interested in staying in the Queen's room is never convinced I'm a safety freak (compared with most audiophiles, anyway, who often ground lift and neutral reverse with enthusiasm).  She's constantly convinced I'm creating some kind of electrical safety hazard, even though the truth is I'm constantly thinking about safety issues that hardly anyone else thinks about.  "Everybody does it" would not be an acceptable alibi if she discovers something arguably substandard about the wiring.  This would prove, finally, that I'm a reckless fool to be watched constantly.  So I do really want to do things "the right way" especially in this room.

I have not had a hum problem with any other of my Sonos connections.  I suspect Sonos uses ethernet transformers for the network connections.  But it occurred to me that a ground loop could be caused by the shielding in my Cat6a STP network wiring.

So I tried isolating the ground by using an ethernet coupler to attach a second piece of unshielded network cable.  Sure enough, that fixed the hum!

(Other "fixes" like using short and stout RCA line cords didn't help.)

Now I had been persuaded that shielded network needs to be shielded everywhere.  And I think that's generally desirable.  But this is only one line that goes straight back to the main fast switch in the kitchen.  I believe it won't cause any harm to the rest of the network to have the very end of this line unshielded and ungrounded.  The ethernet line is way shorter than the maximum run of 330 meters, more like 40 meters.

In fact, I've seen this specifically advised for home networks.  Only shield ethernet cables at one end, some people say.  Others, a whole slew of professionals, recommend avoiding shielded cables altogether, and especially in the home, saying that it's incredibly complicated and difficult to terminate the shielding correctly.  But my thinking now is that the solution is easy.  When ground loops occur, lift the cable shields at one end.

The problem here must be that the Sonos modules wouldn't be correctly designed for use with shielded cables because they have no ground connection themselves.  It's funny, however, that this has never caused a problem before, especially for the last year (that's how long it's been) since I installed an all new home network with all Cat6a STP cables.  When an ethernet shield enters the Sonos chassis it becomes, for all purposes, the effective ground.

I've decided to make the permanent solution a new 10 ft length of unshielded Cat6a cable, from the wall panel to the Sonos box in the queen's room.  Using a coupler is ugly (and the current coupler is especially ugly because it's a Cat6a Shielded coupler, and I think all exposed metal is connected to the shield, making it no good for wet noses.

I don't think the 10 foot length of unshielded cable on one dedicated line will adversely affect my overall network at all.  And the Queen's room is a relatively low RFI/EMI area anyway.  The main purpose of the shielding was to protect the long runs of cable in the attic.

I did also order a 1 foot length of unshielded cable and could use that with a nice unshielded coupler.  That way I could limit the amount of unshielded cable to 1 foot.

Monday, August 25, 2014

EQ thoughts

I'm now leaning toward getting the OpenDRC-DI hardware for room correction, and using Room EQ Wizard to design the filters.  I like the fact that OpenDRC-DI has AES/EBU digital connectors, which can plug right into my existing system, and make the best digital connections over short range.  I like the fact that R.E.W. is an open software product, and I can see what it is doing, and possibly what I would like to change.  The downside of this solution is that I believe it does not allow for multipoint correction.  In contrast, DSPeaker dual core does have a multipoint correction feature, though it is not clear how well it works (and the manual doesn't talk about it much).  And it has only Toslink digital connections, which don't work well with much of my stuff (such as the DCX2496, which has no Toslink input, and my Tact 2.0 RCS, for which the Toslink output seems to be no longer working).

OpenDRC is actually more powerful than some other MiniDSP products.  It has enough power to run both IIR and FIR filters.  REW apparently creates IIR filters, much like old fashioned analog filters.  FIR filters allow for the correction of phase and amplitude separately, which makes it possible to do lots of interesting things, such as remove the time delay variation produced by a speaker crossover, or design a speaker crossover with no group delay.

Sophisticated commercial correction products (Acourate and probably Dirac) use FIR to correct system phase response, but that goes along with doing a full range correction.  I strongly dislike full range correction and want bass EQ only.

http://www.minidsp.com/forum/minidsp-for-newbies/7204-can-i-use-rew-with-opendrc-di-for-roomcorrection-answered


Friday, August 22, 2014

Belden 1695a is Teflon FEP (not the holy grail, Teflon PTFE)

Teflon is an audiophile holy grail.  It has superior dielectric properties to all other plastic dielectrics.

So based on that, I was about to buy a Belden 1695a cable for the final stretch of my SPDIF line from kitchen servers to living room stereo.  Instead of 1694a, which is basically the same, but uses PE foam instead of "teflon."  Even though it's a digital line, the dielectric properties could be important.

But reading the fine print, I see that the Teflon used in 1695a is Teflon FEP.  This is not the Teflon that audiophiles seek out (if they know what they are doing, anyway).  Teflon FEP has dielectric nonlinearities 6 times greater than Teflon PTFE.  Comparing the linearity of FEP with Foamed Polyethylene, I'm not sure which is actually better, but I suspect it might well be the Foamed Polyethylene.  I know that's what my friend Tim thinks.

So I'm getting the cheaper 1694a.

I feel similarly about the FEP used in Valhalla cables.  I don't think much of it.

I wonder about the Teflon used in Cardas cables.

Meanwhile, I've decided to get 1695a to connect the Oppo to the SPDIF panel in the kitchen, because the 1695a is said to be slightly more flexible.  I used 1505F for the line from Mac to SPDIF panel because I thought that needed some more flexibility also.  Now I think I might have used 1695a for that one as well, regarding the solid core 1695a to be the better cable compared with 1505F.

Linkwitz on Room Acoustics

Linkwitz has an in depth discussion of room acoustics.

His take is that generally room acoustic treatments aren't worth bothering with.  Instead, choose ordinary furniture and decorations wisely.  So this is not Ethan Winer's advice…

Generally Linkwitz believes lively rooms sound better.  This is very much like my friend George Louis, who eschews both bass traps and EQ.  OTOH, it is not like my friend Tim, who feels that all reflected sound is distortion.  I believe it is correct that lively rooms, though not too lively, sound better.  Rooms where I have replaced carpet with vinyl plank flooring with acoustical underlayment sound much better for that reason.  Old carpet has an old carpet sound, and it is wonderful to be rid of it.  Vinyl plank flooring with acoustical underlayment is the best I have heard.

For technical reasons, room mode calculation is worthless, and it may not help much to build rooms to acoustic dimensions either.  There are simply too many variables.

The closed box bass radiator design inherently excites room modes several times more than dipolar bass radiator.  Dipole bass should be used when possible.

WRT bass modes, attempts to treat room with absorbers can make only marginal differences.  It is best to attenuate peaks with EQ, but holes cannot be filled in.

Like Winer, Linkwitz also believes that response away from the optimal listening position matters:

The response should not be optimized merely at the listening position. Few commercial products deal with this adequately.

On another page, he praises the Lyngdorf TDA2200 amplifier and correction module, which has correction algorithm by Jan Abildgaard Pedersen--which samples from random points in the room!




What Ethan Winer writes about EQ

Ethan Winer sells bass traps, so not surprisingly he is critical of claims made by sellers of automated EQ systems--who sometimes suggest EQ will handle everything.  I think his criticism is somewhat refreshing in an industry that generally refrains from being critical of any excuse for you to spend money.

But he is also somewhat contradictory.  At the top level he makes exaggerated recommendations like "Just say no to Room EQ."  But when you get down into the fine print, he does suggest that the use of EQ is warranted to tame the 1 or 2 most serious modal peaks.  He uses the EQ feature of his SVS sub to do that.

He also recommends Room EQ Wizard to do measurements and filter calculations, and the use of a $150 Behringer Equalizer to do the corrections.  That's excellent cheapskate advice, straight from Home Theater Shack!

Since EQ can only correct response at one location at the expense of others (a claim that Ethan reiterates a lot…but hardly anyone else in hifi does…especially those who accept the idea that there can only really be only one really good listening position), Ethan suggests only using half as much reduction as measurements indicate.  That's the kind of fudging I've been doing since I started with EQ in 2005.

I continue to do a bit more than just tame the two worst modal peaks (as his recommendations do also), though I've only done manual EQ adjustments, doing measurements with Tact and SPL meters.

The downside of acoustic treatments is different.  You can't make much difference without giving up a lot of wall space and a significant chunk of floor space.  You can spend a huge amount of money to only get a couple dB of difference at modal peaks.  With many rooms, you could line the walls and fill the corners with acoustical treatments, spending $30k or more, and still have bad room modes.  In fact, in most rooms, it's simply impossible to add sufficient bass trapping, let alone too much.  That's why many acoustical absorbers are designed to trap little or no highs…because if you fill the room with high frequency traps the room will sound horribly dull.

In my multipurpose living room I find it hard to imagine where I would put traps to make any significant difference at all.  There is so little room for traps, and the modes so large, I'm strongly tempted to use an active absorber, like the one from Bag End.

I'm currently thinking about a living room redesign to make it better for parties and watching TV.  The result will be even less space available for room treatments.

Thursday, August 21, 2014

Review of Acourate

Here's a great review of the Acourate room and speaker correction system.

The review also shows many other things, such as the calibrated mic kit from iSEMCON, calibrated microphones from Cross-Spectrum, and more.

My collection of calibrated microphones is in bad shape.  I have a calibrated mike for LAUD V3, which works only with that program running on a Win 98 PC with Fiji card.  But I'm unsure that the calibration file is properly loaded.  I haven't run the program in years.

I have two Tact RCS 2.0 preamps which both came with calibrated microphones.  I can't remember which microphone went with which preamp, and the Win7 laptop I run the old 2software on makes it unclear whether mic calibration file is loaded or not (the window is blacked out), or which one is loaded (if I could even remember which was which).  Many Tact users decided that the factory calibration wasn't any good anyway, you'd be better off simply using an "average" calibration file for these microphones (similar to ECM8000) to avoid false correction, especially in the highs.

Somewhere I have a Mighty Mike which was never removed from the box the ebay seller shipped it in.

I have an IVIE IE-30 which I haven't used in a long time and needs battery replacement.  It came with an ACO mike in preamp holder.

I have several Bruel and Kjaer microphone capsules with paper curves made in the 1960's.  They must be use with appropriate microphone holder such as the one on the now non-functional IE-30.  B&K also use very unintuitive way of describing the frequency response/polar characteristics of these microphones.

I have several General Radio microphones which came with my GR1933 SPL meter.  These have even more specialized mounting requirements than the B&K's.  I also have some barely working B&K SPL meters.

I have a pile of Behringer ECM8000 microphones which I got in various ways, and I can't remember which is which.

I have Galaxy meter which was tested by Home Theatre Shack and said to be within parameters of their standard Galaxy correction file.

I can never decide whether the time has come to buy M30 or similar serious measurement microphone, though it could be argued I should have done that a decade ago and avoided some of the other things (especially the Ivie, B&K and GR meters).

At this time, I'm not much adjusting anything but bass anyway, so why do I need microphone with 30kHz response?  Though I would like to measure super tweeters anyway.





HDMI De-Embedder

Here is a great discussion on HDMI De-Embedders to extract high resolution audio from DVD-Audio's and SACD's.

As for now, I get full resolution from the SPDIF output of my BDP-95, and I think SPDIF is likely better than HDMI for transmission anyway.*  I am lucky to have BDP-95, one of the few players to do this.  AND my current matrix switch seems to be blocking high resolution audio through the EDID--I could probably change that somehow, and when I next come across the manual for my matrix switch I will try.  Meanwhile I have the recommended Kanex Pro de-embedder installed, and it works for standard resolution audio.  If I could get HDMI to work for high resolution audio, I could use the SPDIF line exclusively for the Mac connection, and then use a superior F connector cable (see previous post).  But my current thinking is that full resolution through SPDIF is likely better than that though HDMI, so I would be disinclined to use the de-embedder for any other reason than convenience.

(*The HDMI is a 100 foot connection using HDMI to CAT6a conversion.  So it's hardly a "simple" connection either.)

RCA? BNC?

For a while I debated about whether to change the connector of my SPDIF line from kitchen to living room that now carries high rez audio from my Mac/Amarra and BDP-95 to the main system.  This line uses all RCA connectors on the cables now.

This line is particularly complicated for other reasons, when connecting to the Mac.   It works, but I now go through two levels of optical/coax conversion.  First, a very fine Inday Toslink splitter splits the Toslink output from the Mac into 4 Toslink outputs.  This Inday active splitter has always worked perfectly, whereas passive splitters never work.  From the split outputs, one goes to kitchen receiver, another goes to a DAC to provide analog to my hard drive recorder.  The third gets converted from Toslink to Coax via another very fine M-Audio CO2 converter.  As I explained previously, this combination of splitting and subsequent conversion to coax requires two levels of optical conversion (or 4 levels if you count both the optical emitters and the receivers).  It also goes through at least one transformer for the coax (in the CO2).

With all that conversion going on, it's a wonder that it works at all, but in fact it has always worked perfectly, as far as I can tell, though I worry about jitter.  What did not work very long was a different converter I got to split one Toslink to two Toslinks and one Coax.  I had lots of problems with that converter, then after I did get it working, it stopped working after one day.  So back to the Inday Toslink splitter, and the two levels of optical conversion, which bother me but always work perfectly.

Well I figured I could optimize this a bit by trying to make the coax connection better.  Currently that is very complicated.  From the CO2 I run a 12 foot budget video cable from spare box (vinyl cable with stranded wire--about the same as Radio Shack's lowest grade AV cable from 2006) to the kitchen connection panel.  There it runs through an RCA to F adapter, into the panel, where it then runs through 50ft of Belden Precision Video cable.  In the panel in the living room, there is a second RCA to F adapter, followed by two short lengths of Monster Video 3 cable joined with an old RCA barrel adapter.  This is what stuff in the real world looks like.

One way to make the connection better would be to eliminate as many RCA connections as possible, particularly the ones that must go through those questionable impedance RCA-to-F adapters.

The original plan for this SPDIF line was that it would only provide data from the Mac.  Then I would simply get a cable with RCA connector on one side (to connect to the CO2 converter coax output) and F connector on the other side (to connect to the F connector on the kitchen patch panel).  In one fell swoop I would be replacing the low grade 12ft video cable, remove one RCA connector, and remove one RCA-to-F adapter which might be even worse than just one RCA connector.

But once I started feeding the SPDIF line from both Mac and Oppo, I needed a connection which would be easy to change.  Screwing and unscrewing F connectors is a big pain.  For awhile, I looked for SPDIF switches, but there's hardly anything like that available and often it is way overcomplicated with re-clockers and the like.  So back to just using the patch panel as a patch panel, and changing the cables to switch the playing device.  So then back to finding a good connection for patching.

On a lark, I looked for a push-on BNC connector.  Sure enough, Neutrik makes a special push-on BNC connector that was intended for patch panels.  It appears that I could special order a cable with this connector on one end through Markertek, as they are a Neutrik dealer.  The I could get (and already did) a BNC to F adapter for my existing patch panel.

That might be an optimal solution, but I wondered if even a push-on BNC might get a bit bothersome after awhile also.  One thing about RCA connectors is that they ARE easy to plug and unplug.

I also realized I was not seeing the big picture.  I could get just as much benefit by replacing the cable in the living room with one having F connector on one end and RCA on the other.  That would also eliminate the need for the F-to-RCA adapter in the living room, as well as the barrel adapter.  And all this business about impedance matching in the coax line is probably insignificant compared with the two levels of Toslink conversion.

So, for now, I've decided to stick with RCA connectors in the kitchen, where I must change the cables often.  I ordered and have now received from Blue Jeans Cable a 10 foot Belden 1505 with RCA on both ends for the Mac-to-Panel connection.  I used that cable to verify that a 10 foot cable would also be correct for the living room, and I will order that second cable with F connector on one end.

I've got a new plan for the Toslink splitting.  I'll use two (now out of production) M-Audio CO2 converters in series connected together with coax.  It's hard to know if that would actually be better than the current setup though.

Friday, August 15, 2014

Multipoint Room Correction

One of the features I want in my next low frequency room correction system is multipoint correction.  That means that a correction will flatten a primary listing position while at the same time not letting secondary listening positions get too far out of whack.  I need that because the quasi-central listening position gets cancellation but much of the room around the boundary gets huge boom from room modes.

DSPeaker Dual Core has this feature.  One thing I don't like about the DSPeaker, however, is that it only has Toslink digital IO.  The Toslink output of my Tact preamp doesn't seem to work (didn't work with Behringer DEQ 2496 when I tried that a few months ago).  So I will need to convert the Coax SPDIF output to Toslink with an adapter.  Then I will need to convert the Toslink output of the DSPeaker into AES for input to the Behringer DCX 2496 which I may still need for crossover and time delay functions.  I might be able to use the analog output of the DSPeaker if I can program in correct inter channel delay and crossover, and it looks like I might be able to do.

Here is the manual for the DSPeaker Dual Core.

On the other hand, AcourateDRC does not seem to have multipoint room correction.  I was looking at that alternative, as it can run on the MiniDSP OpenDRC-DI, which has AES, SPDIF, and Toslink inputs and outputs (for only $299 w/o software).  MiniDSP claims you can get a license to extract filters from Acourate for only $99 but I have not confirmed that.  It looks like the full version of Acourate is more expensive than that.

Another possibility for the MiniDSP OpenDRC-DI is Dirac.  I haven't yet determined whether it is possible to set an upper frequency limit for the correction by Dirac.

A third possibility for OpenDRC-DI is REW (Room EQ Wizard).



Thursday, August 14, 2014

Switching from Tact to something Else

What's Best Forum addresses the switch from Tact room correction to something else.

A consensus seems to be that room correction is best kept below 250 Hz.  Unfortunately, my earlier Tact 2.0 RCS can only do full range correction.  Later versions of Tact, in a late late update, got a top correction frequency setting.  For the longest time, the designer of Tact maintained that full range correction was essential, so a top frequency setting was not allowed.  That probably doomed the company, which appears to have vaporized (it had a good run for about 10 years).   You could (and I've never bothered to do this) set a correction curve that matched the upper frequency curves of both speakers well enough that there would be little or no correction.  I've never really mastered drawing target curves that well.

That's why I've been using manual room mode correction, for my subwoofer only (except I just last week added a minor notch to the panels).  I'm planning to try the DSPeaker Dual Core correction soon. It apparently has a low top frequency…though it's not clear you can set the correction top frequency.

But even without top frequency for correction, I can fake it by applying correction only to the subwoofer channels (after measurement).  Then whatever correction it might have applied to upper frequencies won't matter much anyway.

Since I upped the bass slightly (reducing the steep notch at 45 Hz) in the last time alignment I have been noticing excess bass boom around the room (but not at listening position).  My plan is to correct both at listening position and at a wall position, so the wall position (with maximum nodes) won't get exaggerated.

Wednesday, August 13, 2014

HDCD described

Here's a good (partial) description of HDCD.

It explains the amplitude processing features:

Peak Extension (up to 6dB of compression, matched by identical expansion during HDCD decoding, as with all other amplitude processing features.)

DSP Gain (+12 to -31.9dB)  If Peak Extension is also used, DSP gain is limited to a maximum of +6dB.

Low Level Extension, in normal and special modes.  Special mode does much more low end compression (raising the lowest amplitudes up to 7.5dB) compared to normal (which raises the lowest levels 4dB).

The post doesn't explain the variable filters, however.

Basically these features let you squeeze about 20bits of resolution/amplitude into 16 bits.

I like HDCD, and HDCD's are some of my best sounding recordings.  It is a bit problematic, however, that without HDCD decoding, what you actually get might have been better without the HDCD (contrary to claims HDCD proponents are always making) because, obviously, you are losing dynamic range if the amplitude processing features are used.

However, the producer might well have chosen to limit dynamic range the same way.  And therein, HDCD offers a quasi dual disc solution.  Non-audiophiles can listen to HDCD undecoded, and get the flatter dynamic range that may be more suitable for casual listening.  Audiophiles can enjoy the full dynamic range with HDCD decoding.

Now that I can play back high resolution files from my hard drive, I'd like to convert HDCD to 20 bits. There are playback software programs (including the standard Windows playback) that will decode, but I'm not aware of any way of saving the decoded data to a new file.



Hi Rez on Oppo BDP-95

I'm very happy I can get 96kHz digital from the SPDIF output of my Oppo BDP-95 playing commercial DVD-Audio discs, like Hotel California and Rumours.  But is it the full 24 bits?  I was thinking I might do some kind of test to figure it out.  Not easy to do for various reasons, especially in the case the output might have 24 bit dither even if not 24 bit information.

Anyway, famous audio reviewer John Gatski has already done the tests.  Not only does the Oppo put out 96kHz, but it's also 24 bit, from the SPDIF output.  He used ATI ASDAC which has bit depth indicator to confirm.  The Oppo BDP-105 does not do this, it dumbs down the SPDIF output to 16/48 like most DVD-Audio players.  So what's with the BDP-95?  I think possibly the DVD-Audio consortium wasn't answering their phone when the BDP-95 was being designed.  But afterwards the RIAA got to them and told them never to do that again.  However even on BDP-105 you can get the full resolution stream from HDMI as well (which doesn't seem to be working in my system, possibly because I have programmed my matrix switch to only allow 2ch audio).

Now even if he used a bit rate meter, it wouldn't prove that the BDP-95 wasn't just dithering to 24 bits. However, if that were true, why didn't Oppo do that in the BDP-105?

So the best information that I have is that I'm getting the highest resolution my system can handle, 24/96, from the SPDIF output on my BDP-95 when I'm playing DVD-Audio.  The best of everything!

The Best of Everything

It's been a harrowing six weeks since I returned from vacation, particularly haggling with the guy I bought a semi-functional Denon DVD-9000 from.  It didn't play most DVD-Audios.  Out of a stack of 8 it played only one.  The seller agreed to pay for repair.  I got estimate, but before I got his approval (he took more than a week) I went ahead and paid to have it fixed.  But replacing the laser didn't help.  In fact, afterwards, it wouldn't play the one DVD-Audio it played before.  Still plays CD's.  So now I was stuck.  I offered either to have the old laser put back in (no guarantee that would restore original mostly broken operation either), take $125 off the refund, or settle for $250 partial refund (out of $699 purchase price) for unit that only plays CD's (and not DVD's).

I might not have offered $250 except I figured that with the $125 off (because the repair made it slightly worse) the amount I was effectively getting off was at least $375 compared with sending the unit back. And if you count the cost of shipping and materials, make that $450.  Then, if you count all my effort and patiently waiting…you see I actually got the player for free compared with sending it back.

Actually, of course, I paid $699+shipping+$89 (for ineffective repair) - $250 partial refund.

Anyway, it rarely happens that I've gotten any such satisfaction in previous cases, so eBay's buyer protection system works (even though I didn't elevate this case to the full eBay buyer protection, the seller was under effective threat that I might).  It's unclear what they would have done in this case.  I might have gotten full refund (but still out shipping and ineffective repair costs).  Or they might have maintained that the ineffective repair made the unit unreturnable.  I think it would have gone my way.

I still look at getting the unit at low cost after some mistakes (like the repair and the minor damage it caused--which I feel correct in taking responsibility for).  And for one thing it does, it might have been worth having at any cost.  For this may well be the Best HDCD Player Ever!

It has the dual differential PCM 1704 for the real PCM conversion.  And the separate power supplies, etc.  And it's built like a tank (which, sorry to say, often has little to do with actual reliability).

Many of the famous HDCD players used predecessors to the PCM 1704, such as the PCM 1702 and PCM 63.

You really want a real PCM converter, I believe, especially for something sampled with a real PCM converter like the Pacific Microsystems Model One and Two.  So you can rule out pretty much all the other universal players that have been made since the DVD-9000, including the estimable Oppo's BDP-95 and BDP-105.

I actually did and A/B/A comparison (I hate doing those) with my DVD-5900.  The difference was as I expected.  Through the DVD-5900, which uses high grade sigma delta conversion chips, the HDCD sounded like you were hearing the music play from inside a jello mold.  The dynamics were inverted, so that transients were convex rather than concave.  The DVD-9000 seems to have measurably higher transient output on HDCD's, maybe by 1-2dB.  I had the change the gain setting on the Lavry to compensate.  I have also worried the extra output might be a sign of some sort of deterioration.  But the DVD-9000 is by far the better sounding.  It's been a revelation like hearing everything the first time.

This is exactly the sort of difference I was expecting from my "information" analysis.  The real PCM player preserves the full information, the sigma delta player shaves the original information off by giving a successive approximation, which only looks good in slow measurements.

So I now have by far the best HDCD playback I've ever heard, and I've been enjoying that on Reference Recordings and an HDCD encoded set of Mannheim Steamroller Fresh Aire I-8.  Which do finally and really sound fresh and not canned as they do on sigma delta HDCD players, or worse without HDCD decoding.

But what about DVD-Audio.  I cherish my DVD-Audio's and in fact just bought a new one (!) this month.  This format isn't dead at all.  Am I still forced to play those through sigma delta derived analog output?

No, and in fact, I can do that even better than I expected, I have found!  My Oppo BDP-95 is the rarest of models that can put 96/24 on the spdif output.  And I can now route that to the living room on coax (as I also do for high res audio on my Mac).  I first tried that last weekend, and was blown away.  It was even better than my DVD-Audio played on Onkyo RDV-1, resampled by Lavry AD10, recorded from digital on a Masterlink, then transferred to my Mac.  This was not a resampling of the DVD-Audio, it was the recording itself!  The bass had even more impact and tunefulness.

So between the DVD-Audio and HDCD, my house was playing wonderfully this weekend, and still.

Actually I don't know what might happen, though, if the DVD-Audio has a 192kHz stereo track.  Does the Oppo digitally convert that to 96kHz?  Then I've got digital conversion…though probably fairly benign.

In addition to the eBay fight I was having (which created a continuous feeling of stress…even going back to before I placed the order as the eBay buy-in-now page malfunctioned and I had to pay from checking, which screwed me up for several days until payday I had to keep adding money to checking), I finally got around to doing the time alignment on my system.  Last time I did a full time alignment it was 3 years ago.  Everything was way out of whack, including one subwoofer being out of polarity (at one time, I was testing that as a way to reduce boom).  The time delays were way off too.  Following the Tact measurements I also tweaked the EQ a bit, reducing the huge notch at 45 Hz a tad, and even adding a slight notch in the Acoustats low bass but well above the crossover point (but nowhere else).  The result measured nicely and sounded wonderful.  The new DAC (Onkyo RDV-1) for the midrange helps greatly, and I'm running the digital typically with little attenuation (maybe 3dB in midrange) due to the low output of the Onkyo, which works beautifully in my system.  This also means, btw, that I can't turn up the volume very much more, though the Tact can play up to +7dB, I'd rather not go above 0.  But having extra volume control range below 0dB which you don't use--is actually a bad thing.

Almost all kinds of attenuation (except amplifier gain switching) are information losing, so having less attenuation also means that much less is being lost.  My typical master gain was more than 10dB lower because the Behringer DCX2496 has 10dB more useless output.  Resistor ladders also reduce information in an analog system--6dB less output from a passive network means 1/2 as much information down to either the noise level (the conventional, but wrong IMO, way of thinking about this) or down to a quantum level (probably around 10-30 below 1V).  Either way of measuring information, you get this loss through attenuation, either resistive or in digital.  (Only HDCD avoids the information loss when going to lower levels, though it's level control features.)

I'm also copying the Reference Recordings HRx files to my Mac and converting them, using Triumph with the Izotope resampler, to 88kHz for playback through Amarra through my digital link to the living room.  They didn't seem to get resampled to 88kHz by the Oppo.  None of my digital stuff goes above 96kHz.  Lavry argued that 192kHz wasn't better anyway.  Anyway, my resampling is "the best" also.