Tuesday, May 23, 2017

Master 7 Triumphant!

The Audio GD Master 7 Singularity arrived on Monday morning, 5 days after being shipped by the factory in China.  It immediately vanquished all my fears and doubts.  Packed perfectly, handled perfectly, and smell free in every level of wrapping.  I didn't recall selecting Air Freight but if that was what this was, I'd do it again.  It was like it was hand carried the whole way, the last step being the delivery into my arms by the DHL Freight Agent.

I let it rest, then unpacked and set it up, then warm up with the power amp unpowered.  Upon unwrapping I immediately had to place it in it's designated location atop the marble slab (where the Audio GD Dac 19 Anniversary Edition had been until minutes before) because there's no other free space large enough in my home to rest it on except the dirty carpet.  When you get the Grand Piano, you have to live around it.  This impressive looking unit is as large as I imagined, but not quite as heavy as I imagined, though certainly no lightweight.  It's large because doing things the right way from end to end takes a lot of well engineered circuits and parts.  To get tiny, you need to make compromises.

Also immediately, it was clear that this was not the kind of problem unit I had feared.  The hoped-for combination of greater sweetness with greater transparency was obvious, though to appreciate it fully took a few hours, listening to many things I couldn't listen to for years, now revealed with such clarity and sweetness it became possible again.  There was never a sense of needing to "turn it off" even playing the unplayable, at very loud levels, and the often more challenging very soft levels.

My audio perfectionist friend agreed that having the voltage set to exactly what's required to drive the power amp to peak power is best, and not waste resolution on useless voltages.  And it seems that Audio GD did exactly as I had ordered, to achieve the optimal spectral balance I needed to raise the subwoofers and super tweeters by 1.5dB, the change from 2.5V max of the old unit to the 3.0V of the new unit.

The slightly astringent quality of the old system with the cheaper DAC is now gone, replaced by endless depth and richness.  What I wasn't expecting was how all the imaging became MUCH more solid and correctly located between and behind the speakers.  That's the virtue of dual mono construction and fully balanced operation and connection!

I haven't done any technical tests, not even measuring the voltage, but I don't think I have to.  It couldn't be THIS good without having nearly unmeasurable distortion, state of the art resolution, and so on.  And it wouldn't have this level of transparency if they had slipped me the NOS version by accident.

Truth be told I don't know how much of the improvement comes simply from using balanced inputs on the Krell, in my system especially.  The Krell isn't powered through the same conditioner as everything else, and though I've never noticed a ground loop, at some level there must be current flow through shield grounds, and the effect of that is virtually eliminated by balanced connections, among their many other advantages.  Mind you, with some equipment in some configurations it is unnecessary and perhaps even suboptimal, but in my system, balanced drive of the power amplifier was a too long overlooked requirement.  (Actually, I tried balanced operation many years ago when it was being power by Behringer DCX crossover, but I feared probably erroneously that it was leading to the excess heating in one channel--the problem that was ultimately fixed by getting the full Capacitor Service from Krell.)

But I also know now that 1704's aren't being used correctly except in differential mode, and better yet parallel differential, as all the big name DAC's from Levinson and others did back in the day they were still being built with the best PCM chip ever.  The potential of this chip is lost without differential operation.

So now, everything is being done right, and it sounds that way.

Sadly for the rest of you, this unit is part of a very limited production using the very last unused Burr Brown 1704's which Audio GD scoured the world for (becoming, ultimately, the biggest name in audio to rely on them, after Lite Audio bailed a few years earlier on making 1704 balanced units).  In future, and for replacements, it looks like 1704 lovers will be scavenging other old salvage units for the super special unobtanium parts.  Eventually, possibly, even better hybrid R2R technology (such as used by MSB) will trickle down to non-stratospheric prices.  Audio GD is trying their hand in that game, but it might be awhile before they can reach the same levels as they had with 1704's, let alone MSB.  Then again, I don't really know if even an MSB Platinum or whatever would sound better than the dual differential 1704's in my Master 7.

Anyway, it's hard to imagine something being even better than this, but that's what I thought about the Audio GD Dac 19.  But I've got what I need for now, I think.  (This was a kind of record setting audiophile purchase for me, I believe the most expensive audio component I've ever purchased brand new, and among things not so qualified only the Krell amplifier cost more.  So I hope it's enough for awhile, the forseeable future.)

I'm connecting the DAC output to the Krell amp with Nordost Baldur audio interconnects I bought awhile back at deep discount, and actually for making the Oppo BDP-95 to Lavry AD10 connection, but I decided I preferred antique Denon players with unbalanced outputs so I haven't needed it since.  The Baldur cables are thin and extremely "fast", which generally means little stored energy.  I don't really care how technically fast the cable is, delaying my experience by a matter of picoseconds is not of any consequence, but that's a good indicator for stored energy because stored energy, as such, is not so easily quantified.  Anyway, these are clearly well made high performance cables that are very transparent sounding, which is what I want.  I'm in no rush to find something better, and they turned out to be exactly the 18 inches required.  Compared to many audiophile cables, they are just good cables, nothing to equalize this way or the other, though you could argue the capacitance is a tad higher than necessary at 20pF / ft.  Belden 1800F does better, at 13 pF/ft.  This is of little consequence for me...you could calculate the low pass for a total of 30pF for my cable vs 20pF for the Belden, into 50k ohm load, with a 10 ohm source impedance.  This will be so high in the Mhz you might think the higher capacitance not such a bad thing at all, but anyway, there's another issue here, and that is simplification.  When they say about interconnects that capacitance is the only thing, they're wrong, it's the only thing that goes seriously wrong in ordinary cables.  In audiophile cables, many other things can go wrong.  But possibly, if nothing is done seriously wrong, additional things be done more right, the things of the highest consequence of all, and that is smearing or not smearing information.  And at that level, dielectric absorption is occurring through complex geometries at every point through the table, and interacting with the complex geometries of magnetic and electric fields.  This may or may not cause excess stored energy at various frequencies.  You could argue...these stored energy effects are at such high frequencies I don't care, I care about capacitance that rolls off at 10kHz.  But, when the capacitance isn't going to cause any rolloff until 5mHz anyway, why not arrange to have it balanced with the electric and magnetic fields not to store energy and therefore smear transients in any way.  Since the smearing can be a nonlinear thing, it can ultimately induce the possibility of something being heard differently, as say some tiny threshold is met either synchronously or not with some other.

The input connection is currently being made through the SPDIF input using two cables and a converter box, the latter being a HOSA which converts AES balanced to SPDIF, and the former being standard Canare and Beldon cables.  This is exactly how I connected the SPDIF input of the Dac 19 which had no AES input.  I've just ordered two appropriate-looking 5 ft AES cables now that I have an AES-input DAC: an Audioquest Cinnamon, and a Geistnote Canare.  The Cinnamon appeals to me because solid core conductors, and everything is silver plated, even the braided shielding.  Silver plating is absolutely what you need to do for best ultrasonic signal transmission, and AES is critically reliant on that.  Professionals may be more interested in the ability to resist stress fatigue of stranded wires.  The Geistnote is a souped up Canare (I got the extra special connectors too) and I like the extra covering which makes the wire run more straightly, which I think is preferable.  I think the Geistnote will be my backup cable, but we'll see.  I looked at all the pro cables and none used silver plating or tinning to protect the copper surfaces.  So, just at that level, I figured universally available Canare basically as good as Gotham, Mogami, or Belden.  As it turns out, I've most often used Canare AES, though my most recent special purchases were Mogami Gold and Geistnote, and I liked the latter the best.  Among the high end cables, Cinnamon is the least expensive with the key features including silver plated solid conductors and shielding.  It's far less expensive than anything else with the key features, and I have seen no other features worth buying.  I trust Cardas construction most of all, and Cardas used to make a 'plain' AES but now only seem to sell the high priced Clear.

AES has gotten a bad rap because of jitter.  Nowadays many will argue even coax spdif is better.  Coax has a peerless ability to transmit high frequencies...hence coax has always been used in radio.  The 110 ohm AES cable is not quite as good as coax.

But, and it's a big but, in a complex system there may be more significant concerns.  Even the tiniest of ground current flows can upset detecting the moment of signal transitions in an unbalanced low voltage signal.  So AES is probably better if you are doing more than just one digital connection.  That was what it was engineered for, and, as it turns out, kinda what I am doing.  AES works well for me with my pro-audio DSP's, sampler and DACs.  And done rightly, the differences in jitter are not going to be a big deal compared to "the ultimate" I2R, which I cannot imagine when I will be using if I ever can.









Saturday, May 20, 2017

10 analog inputs!

My living room equipment rack now has 10 analog inputs, after my having piggy backed the 5 input dB 5 way selector (with Teflon jacks...the upgraded version) on top of the main selector I have been using, recently modified for zero load and zero pass resistance through the tape outputs which I use, the Aragon 24k, which itself has 6 inputs but I must use one to connect the second switch.

Ahh, preamps have never had enough inputs...and now for some time the "preamp" itself has been obsolete or something, though the impressive preamps nevertheless being built approach unimaginability in previous less high high end audio.

Having lots of analog inputs is freedom, just like having analog inputs at all.  It is freedom I cherish.  Digital connections come with policies, DRM, and hidden stuff like jitter and garbage.  While somewhat jitter prone, SPDIF and AES are at least free and open and I like them, but Analog is even better, Analog is free and pure.  Even when generated by digital devices like DVD-Audio players.

Anyway this finally enabled me to make a "permanent" connection of my two tuners to the input hub where the selected analog input gets converted to 24/96 digital by an amazing Lavry AD10.  Everything needs to either be digital or converted to digital to be used in my system, which is based on mind boggling DSP.

Somehow, I don't know, but it seems like liberating the analog signal from various disc players and reconverting that to 24/96 digital sounds better than using the raw digital, in every test I've ever run, and in complete contradiction of (reason?) established practice by virtually everyone else.

Anyway giving the tuners this super high quality AD conversion rather than the pedestrian 16/44 conversion through Sonos, clearly wakes them up, they are sounding fantastically better.  And now I notice finally the Jazz station I've been supporting for years has finally cleaned up the analog distortion they were adding to their signal in large amounts.  I'm loving FM now, more than ever though the 24/96 conversion.  I made numerous previous attempts to do this, btw, which ended in failure.  Back in earlier times, before my rack of last year, back when it was hard to get behind stuff to make even the most basic connections.  (And, btw, a preamp or selector switch amalgamation NEEDS a shelf all it's own, so the "permanent" connections can be constantly revised, as it seems they often are.)  Back then, earlier attempts resulted in motorboating sound from the tuner tuned to the classical station, and just plain distorted (even more) sound from the equally superior tuner tuned to the Jazz station (each connected to the better antennas for each).  But now, apparently with cleaner wiring and the ability to reach behind the tuners and properly adjust their variable outputs is making this work, finally.

I need to have whole house audio based on 24/96, with unlimited analog inputs, and access to libraries and streams (always, all commerical products put digital libraries and streams ahead of analog, at least Sonos grudgingly provides analog inputs, that's why I chose them and have stuck with them so far, but it's been clear for a decade that I need to make my own no-compromises system to do this, and nobody else does).

And yes I think a decent selector or preamp should have 10 unbalanced analog inputs, and at least 5 balanced inputs, balanced output, and simple switched polarity converter taking advantage of balanced output (and assuming the destination device has real balanced input).

There's never enough.  It's always the unconnected things you need to get connected, and the "reference" connected devices you've grown tired of.




Thursday, May 18, 2017

Measurements of a Singularity 19 DAC

These measurements do not look good at all.

When I get my Audio GD Master 7 Singularity, I will be measuring it!

(Previously, I didn't think I needed to, after all, my DAC 19 is perfect, or so it has seemed...)

I'm glad I discovered Super Best Audio Friends also...they're doing a lot of what I hoped to be doing: measuring and publishing.  (It seems more often I do one or the other but not both for some reason.)




Linear vs Minimum Phase

Here's a discussion about linear and minimum phase filters for digital to analog reconstruction.

The admin Ultrabike argues for linear phase...they produce the least phase distortion in the passband. He also argues away the obsession about pre-ringing as nonsense.  I think he makes a pretty good case.  The steepest and most linear phase filter is the most accurate one and the most transparent sounding (described by some as sounding hard, cold, "digital" or whatever).

Of course merely specifying linear phase says nothing about the particular rate or frequency involved.  Linear phase filters can be made to sound relatively warm just like minimum phase or NOS by making them slow.

This is strange, but apparently the more ultrasonics there are, the warmer things sound.  This "warmth" is mostly pseudorandom phase distortion, and that would be no surprise to electronic musicians.

My forthcoming Audio GD Master 7 Singularity Dac will have several Oversampling (OS) options, 8x, 4x, 2x, and (non-OS) NOS, and then combined with the NOS features of the NOS series of DAC's, which seem to include at least two different NOS flavors.  Previously these options were not available in the same DAC unit, but the latest firmware combined with recent hardware upgrades allows them to be.

With all this attention to NOS, I've been a little afraid the designer wasn't focusing enough on the OS designs, but at least he says the OS hasn't changed from the previous OS version(s) of the Master 7, which have been highly praised, so it won't be a step backwards.

From what I've seen above, I'm now not so worried that he doesn't seem to include minimum phase or "anodizing" options in the OS.  Plain old linear phase at 8x OS looks to be as good as anything.

I still wonder if Denon wasn't doing something special also with their AL24, which would appear to be some kind of up sampling, and how that relates to the above.  I'd always thought CD's sounded nice through Denon AL24 machines, even the ones using sigma delta conversion like the DVD 5900, and that was the first player in which I experienced superior sound from resampling the analog outputs rather than passing the digital directly to my DSPs.



Sunday, May 14, 2017

HPM 100

My second unit, not plagued by bad drivers, sounds ok.  My first had two bad drivers, my NOS super tweeter was perfect but my replacement ebay midrange was worse than nothing: scratchy sound (which you can feel, moving the driver by hand).  The availability of even decent looking HPM 100 midranges being something like nil, I decided go try an HPM 700 midrange, said to be physically compatible with the hole anyway, and possibly improved or better.

Many tricks here to get a loud playing speaker.  The first, as always, is not to actually play loud, such as in the bass below 40Hz, where by 33Hz I'm not even sure if there is even "useable" response--it's there but so far rolled off.  Well this bass rolloff means that you don't have to worry about the mega excursions that would otherwise be required to reproduce 30 Hz at 110dB or whatever, let alone 20 Hz.  Actually even by the tuning point around 40 Hz there's already quite a bit of rolloff and around there anyway it isn't the speaker moving anymore it's the port. 

A modern paper purist would argue plain old paper cones, with their natural damping, would be best of all.  True, perhaps, but it wouldn't have that 'monitor' sound.

But I think maybe the midrange could be rethunk, certainly on the first HPM 100 it attempted to cover too far a range, leading to lots of burnout.  The A version limited the range to like a few Hz in the midband, that's actually not such a bad solution, the woofer can support higher, just not quite high enough to crossover with the tweeter.  A modern high efficiency driver might do the improved narrow band even better.

I'll leave my good unit as reference and the other as testbed for new drivers and crossover ideas.  I like the narrow midrange crossover idea actually (contrary to many).




Too Loud

Even the tiniest step above "real" is "too loud."  On first brush, it might seem as things get louder they get more and more immersive, you hear more and more.  But the opposite is true, beyond a point, a point even exceeded periodically in live classical music, the audition system begins to close down, like an armadillo, and one can actually discern less and less, only the stress of loudness.  Which can feel bracing, I suppose, to some.  But it's a low information thing, one big bit.

But I sensed this phenomenon hearing a demo of the Linkwitz Orion system on some fairly pedestrian electronics (22 series Marantz) playing MP3.  It was about 10 dB elevated compared with my loudest listening, 90-100dBc instead of 80-90dBc.  I sensed a loss of information specifically from the high loudness.  Of course the source material probably didn't help.  I wasn't made to feel my going electrostatic was a waste either, though the Linkwitz were clearly top shelf for dynamic speakers.  (Let's say the HPM-100 I bought very cheaply are something altogether different--a chorus of wheezes rattles squeaks and buzzes which somehow combine to form something vaguely carrying the tune of the original.  Update: my second unit has good drivers and sounds ok, and it plays loud, but I have mixed feelings except for my intended purpose as garage speaker.  Anway, There's a strange magic in how louder can sound better louder than softer beyond the point where that happens in other speakers because as they play louder, the stiff drivers become more dynamically responsive to all frequencies impressed on them magnetically, a kind of "opening up" that invites cranking up.)

But I'm getting the same sense (and moreso) from the HPM 100.  In their present evolving condition (quite a bit more wheezing and buzzing than normal, I think, due to somewhat shot drivers if nothing else).  Even ignoring any of the design or maintenance problems of this speaker, which are multifold, I can tell now that there is a too loud, and actually it's not that loud, but more like where I have generally been gravitating too, even with my unwarranted fear of digital gain.

Symphonic music occasionally throws in a sustained section of too loud, perhaps even topped by even louder boom from the drums.  That can be in the 90-100dB range with the final boom hitting 106dB or more.

But it's not always like that, not even mostly.  The median level is more like 82dB.  That's where we get the depth, sweetness, and lyricality.  The loudness elevated sections are to give the stress and passion, or sturm and drang as they say.

It's true, in some kinds of music, rock n roll and derivative and similar popular music is all about the stress and passion.  There is no sweetness.

But since we're making it up anyway, though you could argue a rock concert is "real" it isn't actually any realer than a home or studio performance.  A rock concert is a temporary drunk, listening at home is life.  So it seems to me even the all passion all the time music should remain mostly below 100dB.  And I think I'll be happy in the higher information retrieval range 75-90dB even for Rock.

That's my 5 min. assessment.  And I was going to spend years investigating this.




How I learned to stop worrying and love digital gain

Sometimes, digital sources need to have their volume raised.  This may be less true if you simply have a transport sending digital to a DAC.  But once you have fancy multi-way and multiroom systems combining equipment from many rooms, and analog sampling locked and now unlocked formats from classic drives (I don't know why, but resampling the analog outputs to 24/96 always sounds better than taking the digital stream off the device, even though it goes into a digital connection either way), you often need to do a little boosting, sometimes just to restore original recorded levels, or in the case of home made recordings from analog sources especially, to exceed them a little bit.

But once I begin cranking the digital volume control above unit gain, 0dB, which on the Tact is 93.9, I get a little worried that digital clipping might occur.  This would occur if the digital signal sent from the Tact onward to the DSP processors reaches the clipping point, which is 0dB.  Beyond that point, I didn't know what was going to happen.  Perhaps dropouts from a validity bit being set.  These dropouts could be far more annoying than typical analog clipping, even on a solid state amplifier (and the "sound" of solid state clipping has been rarely heard or correctly described...in a well designed Class AB amplifier it will not be spitting or overly harsh...just a growing soft harshness.  Anyway, not knowing the effect of digital clipping, it might even be some terrible out of band spiking, for example, of the kind that nearly fried my ribbon tweeters when connected to a 250 Amp (that was actually using the 10V RMS analog outputs of a DCX crossover, but the reason why it was outputting so much in the first place I don't remember and feared was in the digital domain).

But anyway, I didn't really know, because I had never tested it well.  And I had another related issue.  Some months ago I discovered Sonos was boosting volume levels so that "max" now had +8dB digital gain, causing premature clipping.  Perhaps this would not affect people using the analog outputs or something, I don't know, but it seemed a misfeature to me.  I took the precaution thenceforth from keeping all Sonos level at -8dB, which seemed to be the new unity gain.  Even fixed output setting seemed to have boost.  Well some time ago I wasn't sure it was still like this...volume levels were seeming softer than before.  I needed to retest.

I'm happy to say that as of today, my Sonos system passes closest to unity gain turned all-the-way up, and on the fixed setting.  It doesn't add extra boost, just the real deal...well almost.  It seems, according to my test, that a 0dB SPL signal is reduced to -1.3dB.  I'm not sure why that is, but 1.3dB of loss of dynamic range is not a huge concern, ,just a minor gripe really, but if I could have my pony I'd have full bit transparency, which I thought Sonos used to have back in the day.  I could be wrong but multiple tests showed the -1.3dB loss with Sonos "fixed" setting (I've set several back to fixed now, no need to keep fiddling to find the most transparent point) on my Behringer DEQ meter, ,and I needed to crank the Tact up to 1.3dB to reach the highest unclipped point, with clipping starting at +1.4dB but barely visible, and so on, at the output of my DAC, which I believe to be faithfully reproducing what it receives from the DEQ, and comparing that with the Triumph analysis of the WAV file.

Anyway, boosted back to normal and fixed, Sonos is now sounding good again.  That's at least two major advances in the past few days since my return from vacation, the first being un-attenuating the Sony 9000ES.

Cranking up the Tact beyond the transposed clipping point at +1.3dB, it simply clipped the voltage rise at that point, flat, pretty much like a transistor amp, with no artifacts at all.  And that continued as I drove it into "hard" clipping, it just made the wave slopes steeper, meeting a flat top at the same point, as I had speculated earlier, boosting the RMS voltage but not actually raising the peak voltage.

Through any of my DEQ units, the clipping would show the effect of the EQ on the waveform as it clipped, looking a it more curvey for example, but still not too bad at all.

Now, sending an unclipped signal at 0dB into a DEQ unit, and then attempting to clip it within the DSP program had a different effect.  Clipping through the DEQ resulted in a compressed output that showed no sign of clipping, just no increase in size.  I'm a bit worried about this, perhaps on real music it would cause peculiar effects more annoying than the slight harshness of peak clipping.  Or perhaps not.  But either way it didn't look any more scary and probably less scary than analog clipping.

So while you can argue that cranking up the digital volume should not be much necessary, better to have the source material reaching peak 0dB or close to it to preserve resolution.  But when you need to crank up the volume, doing so with digital gain isn't any more dangerous than doing it in the analog domain, and probably safer, with regards to level changes that may later cause clipping.

This adds to the case for mapping the output of a DAC driving the amplfier to have exactly the output required to drive that amplifier to clipping, or slightly more, rather than 6dB more or so.  The added affects of digital and analog clipping around the same point should not be bad, if very rarely reached. As I have previously argued, this preserves the dynamic range and resolution better at every level.

Similarly, I've now concluded that adding boost in the DEQ's, all of the DEQ's, not just to compensate but to create more digital gain, is not a good idea, as it shortens the actual digital mapping space just like having a DAC with too much output.  The best gain setting on any digital device is 0dB or as close to it as possible, to preserve dynamic range and resolution.

Pictures later.