Friday, September 30, 2016

Yin and Yang

I'm beginning to fear that the wonderful sounding Pioneer PD-75, the best sounding digital I've ever heard in my home, is adding something.  It's like it's adding extra energy, a fuzz which makes attacks higher and so on.  It seems to be making my Krell work harder too.  This barely registers below 20kHz were it is far far far quieter than the DVP-9000ES playing SACD.  The pioneer shows nothing above -90dB below 15kHz.  However, I fear, a bit, that the rise I see starting at 15kHz continues up for a bit, but my Behringer display stops at 20kHz so I'm not sure how far it goes.  However...the noise is unlikely to go as far as with DSD, precisely because it can't, the aliases would become too big.  So, in the end, some kind of digital filtering must being done, in the case of a 16mhz bitstream pwm, an interesting kind for sure.  Julian Hirsch measured no more than 0.004% distortion 20-20kHz, so the distortion isn't huge, though digital distortion is especially undesireable because it becomes proportionately larger at low levels.  But meanwhile Sony was getting 0.0015% distortion with similar technology, but not sacrificing much to pulse coherency.  Now 0.004% is comparable to many modern time coherent slow filters, while the culty NOS gets you above 1%.

So, that's Yang, extra positive energy radiating out beyond the envelope for the Pioneer--though very little extra, 0.004%, what's with the Yin Sony?

The Sony SACD as implemented in the DVP-9000ES with the same Sony bitstream chips as the SCD-1, which may have been the last bitstream models as bitstream models generate a lot of heat and noise which must be managed, generates quiet, better than 16bit quieting in the midrange, from 50% noise (1-bit 2.8mhz sigma delta operating at no more than 50% duty cycle) by creating holes, holes in time of the stream of pulses, holes and holes made of holes  That's all it can do, the pulses can't get any larger..

It can't get more obvious than that, can it?

Does the "holeyness" of DSD get better in highly oversampled and overquantized sigma delta implementation as nowadays?  Yes, buuut...the holeyness is there, it's made of holeyness, all the blur can't make it go away.

Or, you could just say the Pioneer adds distortion--extra energy, DSD has perfect linearity in principle but ginormous HF noise, which obscures because it's random.

If you can't hear the noise or distortion, does it matter?  Yes, I suspect it may, you may not hear the noise or distortion itself but I suspect it can still add (if it's distortion) to attacks and such, or obscure if it's pure noise.

So the trick of eliminating the need for severe HF filtering is removed in DSD and replaced by huge clouds of noise (which aren't even Gaussian noise, but slightly tonal).  These mostly inaudible but obscuring clouds give DSD it's Yin character, while alias leakage gives a Yang character.

Thursday, September 29, 2016

You Can't Handle The Truth

The Truth is that CD players can correct small reading errors perfectly (these are in categories C1 and C2).  Larger errors (category CU), they conceal by interpolation from the surrounding good data.  Finally, only the largest errors cause the player to skip or stop completely.

The only way I know about to see counts of C1, C2, and CU is to have a Plextor CD ROM drive and the software with it.  The Plextor software had a feature to scan the disc for these errors.  Of course they are also limitations on the part of the cdrom drive hardware to some degree, but if you have a Plextor drive it's probably about the best drive anyway.

Outside of Plextor, no software or device I know of can do this.

What's especially troublesome to me is this "conceal" possibility (and, btw, "conceal" is exactly what it is technically called, I discovered after reading many documents).

I'd like to know whenever my player is "concealing" read errors.  In that case, I'm not getting the "perfect sound" of 16 bits.  I'm getting something just a bit better than garbage, at least for a tiny moment in time.  Listening to a CD, especially a old/dirty/hybrid/poorly-written CD which maybe very important to me, I have no way of knowing if a large portion of what I'm listening to is fill-in.

At that time, it might be a good time to:

a) trash the disc (or return it) and buy a new one
b) try to clean the disc
c) try to clean the laser with a laser cleaning disc (I've seen the turbo ones with teeny fins recommended, and not the brush kind, and never use liquids)
d) service the player, such as by a technician who will disassemble the player, clean the laser with alcohol and check the electronics
e) buy another player, perhaps newer than the antiques I have
f) switch to playing downloads
g) decide never to listen to recorded music again

Some early CD players, notably those using Philips transports and IC's, had a Disc Fault light which would possibly light under CU error conditions.  Or possibly it would take something worse, similar to that which would make the player skip or stop or not play at all (such as if you put the disc in upside down).  I have looked at the manuals of a few CD player models having these lights, and they were not at all clear.  None discussed CD error correction (always perfect) or concealing (always some kind of educated guess).  Such players include:

Cambridge CD3
Marantz CD74

Now it's a big bother to have to watch the screen where the Disc Fault light is during an entire CD.  It would be nicer to have a counter.  Then you could just look when you were taking the disc out and see how many errors there were on THAT disc.  That's what I think all large or reference CD players should have.

Perhaps, however, various companies involved would rather not even have you think about such things.  Perhaps, legitimately perhaps, there would be endless complaints (this machine gives me a CU when my old one didn't) about hardware and software.  Hardware companies would certainly rather not have complaints about their hardware.  And some hardware companies are also major software (in this case, actual factory recorded CD's) owners or sellers, who would get complaints both ways.

And, perhaps legitimately, even CU's are no big deal.  You could probably blind test a disc with 1000 CU's and nobody would tell the difference.  So why am I even worried, when I can "hear" the difference between my old bitstream, my old PCM, and my newer sigma delta?

Well, because I can't be sure of anything I hear, and having an unknown number of errors, any of which might be ameliorated by some corrective action I could take...well, it bothers me as much as anything.  It may not be clearly audible, but the other things I think I hear probably aren't there either.

BTW, after years of cursing DSD/SACD I am delighting in the sound of my newly acquired minty DVP-9000ES playing SACD's, and I was never even aware of how many SACD's I actually have (such as the old RCA Red Seal and Mercury Living Presence hybrids.  Sadly it is one (just one so far) of the Mercury hybrids which won't play on my 9000ES despite new laser.  When then makes you wonder about SACD that play many errors are being concealed?  It is said that the SACD format is automatically concealing because of the nature of sigma delta.  Anything bit lost is no more than any other, no concealing scheme can do much better than that.  I'm thinking there may be some dispersion on the disc also as there is with CD, so the loss caused by a large gap has only a slight effect on a larger area.

Perhaps the format is even more loss proof, like the CDROM and DVD formats, which use an additional layer of correction.

Anyway, I still love CD, concealment and all.  I think mostly people rarely hear the real faults caused by the CD system-as-a-whole when they complain about how a CD sounds.  The system is nearly perfect, extremely transparent, even at 44.1/16.  There are only slight differences you can make while still being faithful to high fidelity reproduction, and not intentional sound doctoring (which is ok too, if that works for you, as it does even for Mark Levinson with his virtual Cello Pallette, or did for Peter Walker--by all accounts one of the greatest audio engineers ever who designed the Quad speakers and amplifiers) which somehow, I never do.

I recall when I used potentiometers to set volume I always had to readjust the balance for every recording.  It never occurred to me that almost all of the balance twiddling I had to do resulted from the non-accurate tracking of the preamp.  Now that was with my otherwise beloved Aragon 28k, because I was way ahead of this as early as 1979 when I had a sealed stepped attenuator, same as was used in the GAS Thaedra.  Every step was perfectly accurate.  I continued to use that in various systems until the 1990's, then began to realize that this was not necessarily the best design, having a 25k impedance, for driving cables.  But, somehow, that didn't sink in and I continued using the volume control of a Citation One preamp, even higher impedance.  Then came the Aragon, and even that, though it measured about the lowest distortion I've ever seen, essentially perfect, using RMAA and my Juli@ card, I always thought it sounded dark.  And the problem with the volume and balance, it was never right!  I think this must not have had P&G as some think.  Anyway, I have loved digital controls from then on, starting with the Classe CP-35 which I strongly and strangely preferred to the Aragon (but unarguably the digital controls had perfect balance tracking, much appreciated), to the digital controls of the Tact which I have used since 2006 mostly as just a digital preamp, what I still feel is one of the most useful things (a big digital selector with volume trim and more) but hardly anyone makes.  It has 0.1dB volume settings, and balance which is never needed, and polarity but it's a bit inconvenient to switch polarity back and forth since they added parametric EQ feature, which I can't use unless I'm using Tact Room Correction, which I mostly haven't because I don't like it messing with the highs, a feature they never dropped from my 2.0 version.  Anway, it could be better, and I need to think about making something like what is needed, but the 2 Tact units I got during the Tact 2.0 factory blowout have served me very well, and almost irreplaceable so, for 10 years.

Anyway, LP's are far worse than that.  But most of the time, with CD's, you just put it on and it sounds perfect, or if not exactly perfect it would not be proveably discernable to anyone.

But, again, that's assuming that there isn't lots and lots of filling going on.  And you never know, because there is no indicator on all but a teeny proportion of players, and even on them there isn't such a convenience as a counter.  And the best of all would be a counter not just for CU but C1 and C2 also, but possibly most systems don't make that information available.  But just along the same lines that audiophiles want a bigger power supply than absolutely necessary, some think having a larger motor drive the CD spindle is better (that's far out, but of course part of getting a PD-75), knowing about C1 and C2 would help you see patterns, trends, etc.  Know if a cleaning might be good.  And, if they even knew, audiophiles would keep those numbers down.

Instead, in at least some cases, they applied CD tweaks which caused more errors or even breakdowns.

But there you go, does the tweak manufacturer want you to know that?  Certainly not, and he is going to badmouth the manufacturer that lets it happen, so a conspiracy of silence ensues.

This company makes a test disc that permits testing of CD player performance in correcting errors.

A good resource on SPDIF is:


I got my Pioneer PD-75, hooked it up, and it sounded so good, I'm not sure I care about the truth anymore.  This is the best sound source I've ever had.  On CD, it totally blows the Sony DVP-9000ES away, perhaps no surprise.  On hybrid SACD's, it depends, sometimes the greater low end authority, wherever that helps such as in Rock or Organ music that's not too abrasive, the Pioneer still wins.  The Sony characteristically has an Etherial sound, which I consider the sonic characteristic of 1-bit systems, including DSD.  Yin.  The Pioneer (which uses bitstream PCM and minimum filtering) has a Yang sound, perhaps even more than slightly macho at times, reveling in shaking the walls.  It does that, and is still the #2 sweetest sounding source in my collection, following the Sony.  It can create sounds I've never heard that still sound correct.

Tuesday, September 27, 2016


Jitter is probably blamed for bad sound too much, when it isn't to blame, and when there isn't actually bad sound.  Jitter as high as 10nS has been shown to be inaudible.

Anyway, I've always wanted to measure it, and now, with my Sencore DA795, I can.

Unfortunately the Sencore doesn't show numerical values below about 200pS, it just says "Low".  Most of what I'm reporting now is just from watching the logarithmic scale meter, which has markings at 100pS and 200pS.  So I'm reporting highly approximate values.

For the Pioneer PD-75, I see about 150-180pS jitter when playing a CD, which falls to 110pS when stopped.  Looks like some room for improvement in the power supply.  However even 180pS is typical for the best equipment ever made.

For my newest Sonos Connect, I measure 180-220pS jitter, clearly a bit higher than the PD-75 and more variable too, measured at coax spdif output.

(BTW, these are the RMS measurements.  The Peak measurements are over 500pS and very unstable, but the Pioneer is about 40pS better there too.  Also these are all unweighted measurements.  I have no idea how the Sencore measurements would differ from those using J-Test for example, but for sure the Sencore is completely insensitive to jitter in the recording.  It is only looking at the clock that can be recovered from the data, not the data itself.  I suspect John Atkinson uses a J-Test derived weighted peak number.  It's beginning to look like my unweighted RMS numbers are lower, but only about 50%.)

In both cases I used 7 foot Belden RG6 with Canare RCA's as interconnect to the meter.

In their original report, Stereophile reported 388pS jitter for the ZP80, which John Atkinson characterized as Excellent.  I thought that they showed 220pS in a later test, but I have been unable to find that.  In the 388pS test he's using J-Test on the analog output, you would think that to be comparable to the digital outputs.

I'm surprised that a heavy old mechanical device like the PD-75 actually beats Sonos on jitter.  But both numbers are excellent actually.  I would begin to feel worried with jitter above 500pS.  Some of the very best audio devices have jitter around 150pS, and often higher.  I'm not sure if I've ever seen jitter below 100pS for an audio device as such, though external clocks can have jitter spec down to 1pS or below.

Well, even assuming my meter to be perfect, there are 3 sources of jitter measured in the timing of the digital output:

1) The clock jitter of the pioneer, and everything that may ultimately affect the clocking out of data at the SPDIF output.  Audiophiles obsess over things like lack of power supply regulation for the transport section (which may be on a different transformer in this unit anyway) affecting the regulation of the clock.  And so on.

2) The cable losses and smearing (I proved this to be a negligible factor by seeing no difference with a far longer cable connected).

3) The jitter made inevitable by the coding of the spdif signal and practical transmitters and receivers.  I do not believe the clock can be perfectly recovered due to the nature of SPDIF data.  That is why the meter specified range only goes to 150pS with SPDIF input.  It goes to 35pS with a pure clock signal at the clock input.

3a) This varies depending on the actual data.  The Dunn test is worst case for 44.1/16.  Most music is less.  A continuing silence would probably be the least jitter, and for that reason I may record a test case.  I love being able to make my own test recordings btw.  Is this possible at low expense with DSD?  I can't make an SACD for 9000ES but I could apparently make a test disc DSD-File or something like that for my BDP-95.

3b) The actual properties of the Pioneer SPDIF transmitter circuit may also be a factor, say compared to a "perfect" implementation, perhaps an over-implementation.  I already set the baseline at "practical" and presumably the Pioneer isn't at that level perfectly, it may not be as good as something could practically get (say, without an Apollo Project--that's impractical for this) in impedance or risetime, for example.  I imagine some obsessing over such things (and even seeing me writing about them think I obsess over them to my casual observers, I merely pointed such things out, unlike say Lampizator I'm only unsure of how important they are in the overall picture, but I'd expect a Pioneer Reference player like the PD-75 to do a pertty good job already at the powering the SPDIF output, and so it seems too, but who knows, it might be 2pS better with a bigger FET running more ten times more current at the output, on it's own separate transformer of course, blah blah, modifier believe they can may everything better).

What is that "minimum" jitter for SPDIF?  I need to read Dunn's articles.

Toslink out!,  AES/EBU vindicated!

I didn't think this was going to be hard.  Several times this week, usually just before going to bed (already way too late), or going to work, I was tempted to hook the jitter meter right at the end of the Glass Fiber Toslink cable I have connecting the Behringer 2496 DEQ to my Audio G_D DAC19.  This would show the evil Toslink jitter, which I didn't imagine to be very bad--especially with such a good cable--and actually the whole system jitter right at the critical point, the DAC for the panel speakers.  And the whole system is what many people consider complicated, many different boxes connected by digital cables that are mostly AES/EBU, including splitters, converters, EQ's, and a digital preamp.

Finally on Saturday night I was determined to do the measurement.  This would be almost worth the price of the meter, it's what I got the meter for!  (Well, actually among many other things...)

And there were SNAFUs right from the start.  It's as if fate itself did not want me to make this measurement.  First, I discovered my DA-795 meter doesn't accept Toslink inputs to the Jitter test.  It's not really clear because it allows "SPDIF 2" and the Toslink input seems to be on the SPDIF 2 side.  I was also maddened by the fact that I couldn't just select Toslink, and I figured that was the problem, that I had selected SPDIF 2 but needed to make some further selection related to the dozen or so icons at the bottom of the page to specifically enable Toslink.  I tried some of those to no avail.  I tried some other test functions and some of them did enable the Toslink input, but you often needed to have an additional input also, such as for the Transparency test where it compares one stream with another.

Finally, I brought up the manual on my computer and sure enough the Toslink jack that was seemingly on the SPDIF 2 side is specifically identified as being a SPDIF 1 input, and the Jitter test ONLY works with SPDIF 2, either coax or AES/EBU.

So the meter maker wimped out!  They didn't even want to try to let you see the Toslink jitter, for possibly the meter might be blamed!

So I determined to find another path.  So I removed the CO2 toslink converter on my Mac Mini along with its AC adapter and a short cable.  I would do the conversion from Toslink to Coax myself, using a converter which has proven itself time and time to be the best, and perfectly reliable.

Fortunately it's much easier to work on this stuff now that the living room floor back around the DAC is at least partly accessible.  But not perfectly accessible, it still requires trick kneeling and leaning on things and so on, all very carefully so as not to damage anything, including especially myself.

But as I was setting up the CO2, every time I nudged the AC power cables toward the wall, I heard something that sounded very weakly like sparking.  Then I just rotated the big 3-way adapter plug in the lower position of the super heavy duty Pass & Seymour Spec Grade outlet, and my UPS started running on backup power.  I played with this only a second time before panicked, I shut the digital equipment off, I shut the UPS off, and I unplugged the big 3-way adapter plug from the wall.  My girl friend had asked me if the big 3-way adapter was OK at least once.  I had actually asked my electrician about changing the two outlets to 4 outlets and he said it was possible but that was months ago.  An alternative needed to be created Right Now!

So this was another emergency project.  I would not operate my system again until it was safe!  I decided to plug the UPS straight into the wall, then plug the subwoofers into the TV power strip which was right there, and unplug the TV and a TV adapter to make room for those plugs (there are no spare outlets in any of the strips--this is the usual situation).

Now by this time the Jitter Meter had clicked off because of low battery, so I had to put it back on charging for a half hour.  Another delay.  But I was determined!!!

So then a half hour later I see that the Jitter Meter doesn't show any battery indicator like it's supposed to, but it runs OK on battery power, so I'm finally able to set up my test.

And I'm not getting any digital lock!  I reconnect wires, fiddle about, nothing.  Finally I notice once and awhile the jitter arrow shows up at about 500nS for a split second, then disappears.

So then I tried using coax instead of Toslink, but this was complicated because the DEQ's only have Toslink and AES outputs, nothing for Coax, and I haven't seen my AES-to-Coax adapter in years.  So I reclaim the adapter that was currently (and now previously) in use creating the coax line for the supertweeters.  I connected that to the midrange DEQ, and got a short coax cable to measure the jitter at the end of.

So now finally I'm getting lock and jitter measurements.  What at first I see is a bit troubling, the jitter playing a CD on the Pioneer PD-75 being redigitized to 24/96 by the Lavry and through my system (I figured the Lavry clock was about as good as it gets...and that's basically driving all the downstream clocks, so this should be nearly perfect and it has always sounded wonderful) was showing around 500pS jitter.

Then I try switching the digital input to Sonos, and I'm seeing only 280pS jitter.  That's not to bad for a whole pile of boxes connected to Sonos which itself measured about 220pS jitter.  All that extra stuff is only adding about 60pS more jitter.

So then I reset the Lavry to 44.1kHz, and the jitter is now only 290pS, just a tad worse than Sonos.

Playinng non-silence through the Lavry at 96kHz was showing about 380pS jitter.

I find this all to be quite signal dependent.  Digital silence is low jitter, classical music medium, hard rock somewhat higher, and background white noise (the -118dB noise of the Lavry itself) reads very high jitter like that 500pS I saw originally at 24/96.

Perhaps the Lavry show was showing a tad more jitter at 44.1 because the program was slightly different, or because the Lavry itself uses the lower bits more effectively.

Anyway, I'm not going to let higher jitter dissuade me from using High Rez!  I think the Rez is more imporant than 100pS more jitter, which may be averaged out by the DAC anyway.

Also, I'm not going to be planning to use Toslink any more.  My very complex system yields what I now consider surprisingly low jitter using only coax and mostly AES.  Only a tad more jitter is added because of all the complexity of AES DSP units, mostly the jitter seen is mostly only what was present at the source.

All my AES splitters and converters have now vindicated themselves, and I'm getting a second HOSA AES to Coax converter.  Meanwhile, the one unit I have is working for the panels and I've run Toslink to the supertweeter DAC which is less critical I think (can hardly be heard anyway).

My high rez line from the kitchen has always used 2-stage Toslink reconversion, because that is the only practical way to run this from my Mac to several devices.  I wrote about this here years ago.  Now I don't consider that suitable for high rez and I've ordered a New In Box Logitech Transporter.

When I saw the review for the Transporter in 2009 I regretted having gone with Sonos, but still at $1999 it was too expensive I thought and I didn't need much of the functionality, such as the analog outputs, and I would like other functionality, such as analog inputs to be available on the network as Sonos does.

I've had alternatives for handling high resolution discs, but my alternative for high rez files was through the Mac Mini and the dual toslink connection, which I now know needs to be avoided.

With the Transporter I get local control with a remote, no need to buy and maintain a separate computer, and now at the $588 ebay price reasonable.   A more up-to-date system like Sonore would require a new USB DAC...  But I don't want USB!  Or a computer in my living room, constantly crying for updates and maintenance.

Transporter is essentially a little stripped down AV computer which can do a bunch of things, including selecting the music to play, playing it, and putting out analog or digital in AES or Coax.

Of course it won't do DSD (directly anyway, without buying another gizmo, a DSD DAC which can accept DSD over SPDIF).  I don't care about DSD64 files anyway, I don't think I'd ever want any.  But DSD128 has some promise.  I could get a Pono to play those, and get them redigitized through analog at 96kHz, which is basically what I have to do with all DSD files since I have DSP based crossovers.  Maybe tomorrow.  I had been thinking about getting a Pono all week, but the Logitech was more mandatory IMO.

Friday, September 23, 2016

More on DSD vs PCM

Here is the most helpful and informative page I have ever read about DSD, by Charles Hansen of Ayre Acoustics.  DSD ("Direct Stream Digital") is simply a meaningless trademark term which Sony has in this case defined as 1-bit delta sigma modulation at 2.8224Mhz and 7th order noise shaping.  They own the trademark so they can say it means anything they like.  Any time you deviate from 1-bit, as is essential for any kind of mixing or mastering or even level setting, you are forced out of the delta sigma domain into the PCM domain.  The Sonoma so-called DSD workstation is really a PCM workstation that happens to operate at 2.8224Mhz with 8 bit data.  DSD and PCM are interpreted by the same delta sigma DACs just with different digital filter algorithms.  The difference in filters explains everything people hear--it has to because there are no other differences.  Any superiority comes from the loss of the need for brick wall filters in high speed systems.  Now that we have 4x PCM, we don't need brick wall filters in PCM any more either, so we can achieve the same benefits with PCM which is far easier to work with, but few have ventured into this new landscape yet (except Ayre of course, who has a QA-9 A/D converter with no brick wall filter, instead it uses a "moving average" filter which has not time smear or ringing).  The only "pure" DSD recordings are all analog then converted to DSD, or live performances.  And there are just a very small number of those.  [And btw, Charles Hansen is the greatest!!!  After reading this hugely informative yet no-nonsense post, I'm a fan.]

Here's a long discussion at Steve Hoffman, which features fans on both sides, and reasonable civility.

Lotsa people think DSD and HighRes PCM are pretty equivalent.  I think that's a reasonable view, though one I don't exactly agree with (I still favor PCM), with the equivalence being DSD is about the same as 88.2/20 or 96/20.

Of course, DSD fanboys have always claimed that DSD is some special magic, that NO PCM could equal.  (Don't tell them about the feedback that delta sigma systems rely on.  That might collapse the magic.)

Many if not most think 44.kHz/16 bit "perfect sound forever" is still perfectly fine, and I know a number of people who think CD quality PCM is superior to DSD, especially in the highs, with a common thread that the highs in DSD sound fake, for which there is a tiny bit of technical justification (more noise and noise shaping is going on, meanwhile there is less brick-wall filtering, so you could also take this the other way).

Famously much SACD and DSD content is made from PCM sources, often defeating one of the long running claims (which probably most serious audio engineers would regard as hype) that DSD bypasses several phases of processing used in PCM.

Although unleashing DSD onto the world, Sony supported it poorly according to many industry insiders.  AFAIK, and in contrast many earlier format releases, Sony did not sell any DSD mastering equipment.  Instead, they gave it away to specific "partners."  If you were not one of the handful of chosen, you were out of luck, you would have to do your mastering in high rez PCM and convert to PCM.  Even the equipment Sony gave away may not have been fully featured, apparently Sony designed a fully featured DSD mixing system with a European partner, then never actually bothered to make it.  It's not impossible to make such a thing, and I believe that there are now, 18 years after launch, fully DSD mixing and EQ system(s) available from companies other than Sony.

Speaking of how DSD allegedly bypasses the decimation and integration phases (the hype which some believe as the magic of DSD that makes it inherently better), there are a bunch of problems with the argument (in addition to the one that PCM processing is nearly always used anyway).  Even if you had pure DSD mastering and playback (almost never the case) the claim would be inaccurate because:

1) First, it assumes that 1-bit DAC's are being used at the DSD sampling rate.  This is almost never the case anymore.  Almost all DAC's used for DSD now are Delta Sigma DAC's.  It's still considered DSD if you use Multi-bit Delta Sigma DAC's at the DSD frequency or higher, which requires a lot of complex mathematics to do  optimally.

The last 1-bit DAC's were used in devices like my 2001 DVP-9000ES.  Those were Sony DACs which actually operated at 70Mhz if I understand correctly, which would be something like 24x DSD.  Sony was doing some kind of way up sampling to increase dynamic range.  So it was never as simple as the cute block diagram Sony used to make DSD look simpler.

(Interestingly enough, it does not appear that the spec sheets for the Sony converter chips used in DVP-9000ES and back to CDP-707ES, have ever been made public.  But Sony did advertise these as 70 Mhz 1-bit converters.  I wonder if Sony made these at the long closed Sony Semiconductor factory in San Antonio, Texas.  Sony subsequently found it cheaper to buy off the shelf multibit sigma delta converters from the likes of Burr Brown.  Cynics

2) Second it assumes that 1-bit Sigma Delta ADC's are used.  I haven't found much discussion about this, but I believe that in the early days of digital audio, sigma delta ADC's were considered too noisy.  Noiseshaping is required when you use a sigma delta ADC.   Also, very high oversampling.  I believe some if not all of the earliest ADC's were actually SAR (successive approximation) which is one of most widely used approaches for analog to digital conversion.   Even now when Sigma Delta ADC's are used, they are used with multi bit converters and high oversampling.

Even if Delta Sigma ADC's are used, there's a lot more going on than you might think.  Quoting from the above linked article:

These are usually very-high-order sigma-delta modulators (for example, 4th-order or higher), incorporating a multibit ADC and multibit feedback DAC. 
Sigma Delta systems are inherently approximate (aka noisy) systems which almost always require feedback to operate correctly.  This is something NEVER mentioned.  This is one reason why I've personally moved back to PCM as much as possible.  PCM does not require feedback to work correctly.  The dirty word "feedback" would destroy the claimed "magic" of simplicity.

Of course it is also because of feedback that delta sigma systems can get near perfect linearity without requiring extensive trimming the way PCM systems do.  You do the fine tuning before, when you can only guess, or you do the fine tuning after the fact, which can always be perfect.

Now this also is probably a non-issue.  While the feedback used in Delta Sigma would smear the highs, Delta Sigma ADC's and DAC's generally operate at such high frequencies that high frequency information might even be better preserved as compared with slower PCM systems.  It's actually quite hard to know without extensive analysis and/or testing which system preserves the high frequency integrity better.

However, one can also just look at the measured performance.  DSD does quite well compared to 16 bit systems in the midrange, but has much more noise in the upper octave 10-20kHz.  That greater high frequency noise means that by definition high frequency information is NOT being preserved as well.  OTOH, there is ultimately response to an even higher frequency, and there may be less phase shift in the upper audible octave.  So it looks like a toss up.

Listening Tests

The best published investigation of audible differences between PCM and DSD was done in Germany using some of the very best megabuck PCM and DSD equipment.  (IIRC the PCM was either 88.2kHz/20 or 96kHz/20, so as to have comparable bandwidth and bit depth.)  Monitoring was done with Stax headphones (you can't get more transparent than that).  And the result was: there is no audible difference!  Not only was the null hypothesis not rejected but most identification was no better than random for nearly all people.

I believe this is basically correct.  DSD is simply an inefficient high resolution system which takes more bits to achieve 88.2kHz/20bit fidelity than PCM does, and PCM is more easily worked with in many ways, including incrementally increasing fidelity with just a few more bits.  The very idea of 2xDSD and 8xDSD is monstrous--a monstrous waste of bits.

I've argued that DSD operates a bit as if it has an infinitely varying digital filter.  Varying the digital filters in 44.1/16 can make a slight audible difference (or larger if you throw out the book with NOS, which is not high fidelity IMO).  Once you get to modern apodizing reconstruction filters using ordinary PCM, it's not clear from published research that better can be achieved or is necessary, but an end-to-end apodizing system like MQA promises to be would be a step better.  That is, a step better than DSD-in-principle.

To DSD or not to DSD

If Only Sony had marketed DSD on a fairly straightforward technical basis, I might have signed on in 1999 and never looked back.  Forget the simplicity crapola, the real technical advantage of DSD compared to plain vanilla PCM is the superb impulse response.

[Update: after the nth revision of this post, I discovered that Charles Hansen had already debunked the above graph in great detail.  It's a pack of lies from beginning to end!  It's no wonder that Sony didn't plaster this on everything, more people might have called them out.  This is not to say that you couldn't come up with a relatively more honest graph to make the point that DSD has better impulse response that the usual 44.1kHz plus brick wall filtering usually used, but in that case there would be competing hirez PCM systems that could do as well.  The way the graph is shown no real systems can produce those results at all.  BTW, I'm now a little bit concerned that the MQA impulse response graph shown in TAS is also inaccurate, though in showing more rather than less time smearing with standard PCM.]

Now, PCM defenders will argue, and they have got to be at least mostly correct, that this difference, which is caused by high frequency phase response in the anti-aliasing and reconstruction filters, is not audible.  But it sure looks like it would be important.

I had hunted all around for a clear picture like the above, and found it posted by Hiro, a senior member of ComputerAudiophile, at this 64 page mega argument about DSD, which looks to be one of the better discussions on the topic.

Hiro starts by calling most of John Siau's arguments as wrong or misleading, then makes a pretty interesting (and misleading) argument himself.  He claims that Archimago measured the same noise from DSD64 as from 192/24 on the Teac interface.  So I had to go and read Archimago's blog.

Hiro is plain WRONG!

In this page of measurements, Archimago shows the astoundingly high noise level of plain DSD in a scope trace of a 1kHz sine wave.  Then, later on, in the 6th graphic on the page he shows the noise level of the Teac under several conditions, DSD with FIR1-4, and 24/192 with sharp filter.  These could hardly be more different above 20k.  The DSD with all 4 possible FIR filters rises from -145dB to about -75dB around 90kHz, reaching -100dB at 40kHz.  Meanwhile, 24/192 rises from -145dB at 20khz to -115dB at 90kHz, reaching only -140dB at 40kHz.  At 40kHz, the point where Hiro claimed that the noise levels were still the same, there is actually a 40dB difference.

Now, in the previous page of measurements, Archimago shows that 24/192 is noisier than 24/96 in the Teac and this is typical of all DAC's (and part of the reason why Dan Lavry and some others recommend 24/96 instead of 24/192 for the highest fidelity).  Perhaps it is not surprising that Hiro takes the worst case for PCM noise, 24/192 as his basis of comparison with DSD.  But even there he is wrong, as I just reported.

Even if you take the worst case for PCM noise, 24/192, and then combine it with No Oversampling (NOS) which as I always argue isn't really high fidelity or standard PCM, do you get the noise to rise a bit closer to DSD.  But the DSD noise is still higher.  Archimago doesn't show the NOS and DSD noise spectra on the same graph, or even the same page, but I can compare them and they are still quite different.  At 40kHz, 24/192 with NOS reaches -113dB.  Meanwhile, DSD64 has reached -100dB, which is 13dB worse.

Now Hiro was wrong in what he said.  But perhaps he merely misspoke.  Perhaps he meant to say that DSD128 is comparable to 24/192 in noise level.  And for that comparison, Archimago does show both on one graph.  At 45kHz, actually pretty much everywhere, the noise level for 24/192 is lower, but just barely until 50kHz and above.  At 40kHz it breaks down like this:

24/192/sharp   -139dB
DSD/FIR2      -137dB
DSD/FIR1      -131dB

I still wouldn't say "they are the same below 45kHz" but close.  But I'm not even sure why we are doing THIS comparison.  Well Hiro also mentions that DSD64 can very simply be up sampled to DSD128.  Now here we have an interesting case, however.  Upsampling will push the digital noise upwards.  But it seems to me very much unlike "noise shaping" in one critical way.  As a purely 1-way process, up sampling cannot possibly restore lost information.  The information loss from the original DSD64 encoding cannot be undone.  So while the noise will be reduced, the lost information cannot be restored, and I'd predict a kind of dark sound, the same thing you get in clunkier fashion with noise gating.

Meanwhile, I would have been (and was) rightfully turned off by a large number of things about DSD right from the start:

1) DSD recorders have been almost unobtainable (there were no consumer DSD recorders until 2007 or so, right now one is available for $999).
2) SACD discs are impossible for most people to make, they require a manufactured watermark (some old machines will accept a fake DVD/SACD, and the newest ones will read DSD files).
3) DSD does not lend itself to simple DSP for crossover and room correction functions--so conversion to and from PCM is required anyway, so the best approach is high rez PCM end-to-end.

I'm less bothered by (3) than I was years ago, for an interesting reason.  The reason is that conversion to and from PCM is extremely transparent.  It's so transparent that I find I often prefer taking the analog outputs of digital devices and resampling to digital at 24/96 than just letting the 44.1/16 pass through all the way.  So, if I'm fine with resampling analog to PCM, why not DSD to PCM, or even DSD to Analog to PCM?  I see now I can fit DSD into my system as a perfectly fine music delivery system, though not as a final digital conversion approach.

Of course, as many have pointed out, SACD was an attempt to impose DRM on an industry.  If Sony could have led everyone to abandon PCM, we'd be locked into their new system with a DRM system that has not even yet been broken.  Of course, we know in retrospect this was never going to happen.

But from the beginning, there was no consumer recording of the new formats, and, very curiously, the first generation SACD machines had problems dealing with user-recorded media that had already become well established by that time.  As if to send a message to the industry.  Well it was too late.

Now I just said that PCM conversion is very transparent, as was demonstrated by the Meyer/Moran experiments in 2006, up to 10 levels of PCM conversion/deconversion was still found to be audibly transparent.  Very little noise is added, however there is a increasing amount of high frequency phase shift.  This doesn't look good on photos but has never been proven to be audible.

Meanwhile, DSD is not amenable to multiple generations because of high frequency noise that keeps on growing until you get overloading in the highs.

However, DSD128 is looking like it might have reasonably low noise levels in the near supersonic, and still of course give you the natural (noise shaping aka feedback driven) impulse response.  DSD64 is so noisy you can easily see the HF noise on high bandwidth oscilloscope traces of sine waves, as Archimago shows.  DSD128 looks just like analog on the scope.

I'm not sure we've seen the end of this, since now filter designers are showing how perfect impulse response can be obtained with PCM and slightly higher sampling and end-to-end mathematical apodizing.  This retains the advantages of PCM in relative compactness and mathematical tractibility--it can easily be worked with in DSP.

DSD counfounds mathematics not because of sigma delta itself--that's the trivial part that had me fooled for the longest time.  Equally fundamental to DSD is Noise Shaping, based on continuous high level feedback.  This means, in effect, each pulse is NOT equal.  Each pulse is in the context of everything before and after it, which actually determines what it means.  This context dependence makes the mathematics infinite.  You can't just "add things up" to make a mixer, etc.

Meanwhile, PCM is reborn every few years with some interesting innovations, though I consider apodizing important but little else.  IMO, by the time we get to CD players like the Pioneer PD-75 around 1991 we're in the modern era of high sound quality, thanks to high linearity, low jitter and high stability, and flat but closer to linear phase digital filters: plain old 44.1/16 bits done fairly well is incredibly good!  For the longest time, the best published research in JAES was that it was perfectly transparent.  It may never have been perfectly transparent, but it's obviously quite close.  It has only barely been established in the AES literature that it isn't perfectly transparent, that significantly improved apodizing can be slightly audibly better and demonstrated in DBT (published by Meridian).  This has not been scientifically established for DSD, in fact the reverse has been demonstrated in the most recent and well done experiment--it is indistinguishable from comparable PCM.  Most talk to the contrary has not been well founded.

DSD stays alive simply by slowly making what used to be impossible less so.  And I'm happy to play with it as I can without huge expense.  I will never have full DSD end-to-end because that would require me to give up DSP.  But I can accept DSD inputs, converted through analog resampling to 96/24.

Which in a way, is not surprising.  Recall that DSD was originally invented, in the first place, not as a mixing or mastering format, but as an archival format.  Now I'm not sure it's especially good at that either, because of noise, but for an archival format there isn't much concern regarding mixing and mastering, and even distribution and playback.  Also, DSD56 was invented specifically for the mastering of 44.1 and 48kHz, the two popular rates of the time, but not for the high rez PCM formats of today.

Now certainly someone as astute as Ted Smith would understand the practical and mathematical difficulties of DSD.  Nevertheless, he built a DSD DAC.  Maybe he has some answers to the other problems too.  I find his progression from first time electronic builder to advanced DAC builder unbelievable.  In this story line, it all happens in a few months in 2010, while he's apparently also listening to Johnny Cash.

Archimago does usefully propose combining DSD128 with lossless compression.  If it can indeed be compressed to the same size as 192/24, perhaps it's not that bad.  But we have no reason to believe this complexity is needed.  As far as we know now, 24/96 PCM is as high definition as is needed, and it is far easier to work with than DSD128.

Here's a comment by Charles Hanson saying flat out that DSD is unnecessary (because he has incorporated the goodness of DSD into his quad rate PCM with the QA-9).  He says his quad rate PCM sounds better than DSD at any rate.  I believe him.  Hanson gives the background story of DSD vs DVD-Audio in a subsequent comment.

Tuesday, September 20, 2016

Clocks, clocks

We can tell the Pioneer PD-75 and PD-95 clocks are about the same by reading the modification instructions by Octave who makes clock upgrades.  The replacement instructions regarding the parts changes are identical.

So the PD-75 has the same clock circuit as the PD-95.  And BTW it's not the cheapest circuit, it's the step-up circuit.  In the cheapest circuit a crystal is simply connected to the relevant chip with suitable passive parts.  In the step-up circuit, the crystal and passive parts are buffered, in the PD-75/95 by two layers of opamp buffers.  An initial layer which then drives 3 separate buffers to send clocks to different things.

Not to say, there are a bunch of more sophisticated designs than the first step up.

But I think it's clear that these were intended to be in the elite upper category of CD clock performance (there were two categories established by the Red Book).  So we could expect perhaps jitter at least to be less than 300pS, allegedly 1/30 or less of what would be audible, or less.

The PD-95 might benefit even in jitter performance simply by having better power supply.  That has huge effect on simple oscillator clocks.  Good clocks are designed to be immune, though nothing is ever entirely immune from influence, within the same electronics.

Meanwhile it's bugging me that nobody talks about simply replacing the Sonos Connect (formerly known as Zoneplayer ZP80 and ZP90) clock the way one might do for a CD player.  (BTW, in some accounts replacing the clock is essentially the only thing that ultimately determines player-added jitter at the output terminals.  The clock ultimately controls the clocking out of bits, and it doesn't have to vary that rate one iota because servos earlier in the system keep buffers filled with enough data.  If not, there's a skip, which rarely happens anymore with good players and discs.  In other circles, everything counts.)  Instead, it seems the major recommended replacement now adds a whole circuit board which does reclocking on the output.  I would rather avoid reclocking because it only just smoothing the underlying jitter.  It's not really eliminating variation because it can't, it's shifting it somewhat to a lower frequency, so jitter becomes very low frequency wow.  And not only reclocking, but ASRC to 96kHz, or your frequency of choice.  That might actually be good for me, but I think I'd rather do analog resampling than ASRC.  And I've had a longstanding bias against ASRC as simply "buring the jitter in the data, where it cannot be removed".  So why not just replace the clock.  The blurbs all say "multiple lousy clocks."  I'd like to see that for myself, I still think fixing the actual clock is better than any band-aid ASRC.  If it's true there are multiple clocks, why not replace the one that counts most, or all of them, or something?

I suspect that the difference between the existing PD-75 clock and something better would not be by anyone audible in ABX DBT.  But of course, I gotta have the best clock anyway, just to be absolutely sure.

I've asked Kingwa if he still makes his Audio G_D jclock.

Monday, September 19, 2016

Replacement for Sonos

Something like this would be a start, the RoonReady Sonore Sonicorbiter SE.

Article also mentions a bunch of network audio frameworks:

HQ Player NAA

Sonos recently has made it hard to select 0dB on a Connect so you can properly adjust level on actual system.  (Sonos is more and more thinking of connecting to all their own devices, rather than serving as a transport layer for music for all devices.  A slippery slope going the wrong way IMO.)

Sonos has never supported Hi Rez digital, or DSD, and probably never will.

Sonos gives no ability to adjust buffering size (and conversely latency) when sharing analog sources over the network.  It seems to be limited due to small buffering to a maximum of 5 uncompressed analog source connections on a Sonos network.

The one good thing that Sonos does that nobody else does is allowing analog connections across the network in the first place.  I cannot switch to any other system until it provides similar or superior analog source connectivity

You might notice this little Sonore device has no analog inputs.

Article comments also mention some other devices:

Raspberri Pi Audio

In this article about a modded Connect, yet more things are listed:

Denon's Heos Link
AURALiC Aries/Mini

BTW, I don't think RE-clocking and up sampling Sonos output to 96kHz is necessarily the way to go.

Reclocking in general is band-aids, all that re-clocking can ever do is low pass filter the timing variations.

What is needed is much simpler: add a word clock input so external clock can be used!  (OR, bolt high quality clock inside.)

Sunday, September 18, 2016


I'm very much enjoying the sound of my new DVP-9000ES playing SACD's.  I'm feeling that, indeed, Sony did something special, at least with these early SACD players, that I have not heard in non-Sony branded players.

Something different anyway, perhaps not all to the good I'm wondering.  Nowadays even Sony uses highly oversampled multibit DAC chips, sometimes even from the likes of Burr Brown, to implement SACD, just like everyone else.  Back when the first SACD players were introduced, Sony was using their Pulse Converter chips, CXA8042AS.  Those pulse converter chips are used in both the SCD-1/777 and the DVP-9000ES.  Outside of that, and the use of one OPA213, the output circuits are quite different between the super high end and merely upscale, with the super high end showing far more additional stuff, and discrete circuits in the output.

A leading audio engineer, Stanley Lipschutz, and several others, revealed inherent faults in 1 bit delta sigma in AES papers in 2000, just after the public release of the SCD-1.  The principle fault with 1 bit delta sigma is that the background noise isn't gaussian, it's tonal, with idle tones.

Sony then denied, to John Atkinson, that Sony was using 1 bit conversion, in response to a paper by David Rich describing the concerns of Lipschutz and others, published (of all places) in Stereophile.

I'd always wondered if that denial was with regards to future SACD, not past.  In the past machinery, as in all the machines I just mentioned, still for sale but designed years earlier, they might have used 1-bit, but that was now water under the bridge.

I don't know enough about these CXA8042AS chips actually to confirm my version of this, that Sony abandoned true 1-bit when it became apparent that it didn't down scale well, or they hit a brick wall in making further improvements, or were covering up faults all along, or were embarassed by Lipschutz, or something like that.  They had been able to achieve brilliant sound in the 3 early players by perhaps mixing things up a bit (especially in the SCD-1/777ES) so that the 1-bit fundamental character was somewhat obscured.  I think they may have been further oversampling the 1-bit, IIRC 70Mhz was claimed for an earlier Sony ES CD player.  But in this regards, the 9000ES is actually pretty straightforward in the analog section, aka simple, and in that regards very different from SCD-1/777ES, but neverthess (or alternately) similarly good sounding.

The alternative is they never did anything like true 1-bit all along, but that seems to be rewriting history.  1-bit arrived on the scene I think starting with Pioneer machines, and perhaps others, in the late 1980's with great fanfare.  One trademark name was Bitstream, another was BASH.  It may have been Sony, who had been plodding along with Philips chips, that was the follower, coming up with their own 1-bit chips to compete with Pioneer in pushing linearity beyond -80dB down to -100dB and maybe even -110dB and beyond.  So the 1-bit race kept on during the 90's, with Sony ultimately achieving within 1dB of the theoretical possible CD THD+N performance with their 997ES and possibly 707ES and pusing linearity out to -110dB.  Pioneer missed this considerably in their PD-75 but I'm not sure about the higher end ones.

But already, by the time say of the PD-S06, Pioneer seems to have been questioning 1-bit.  That machine and an increasing number of later machines, switched to using full width multibit PCM chips like PCM56 and PCM63 especially.  Or at least they were letting customers have a choice, as early as with the PD-93--which used PCM63, but I think Pioneer had completely abandoned 1-bit by the time of the Lipschutz papers.

But Sony had already tied its hands to 1-bit distribution (if not implementation) with SACD, and kept promoting 1-bit until the launch of SACD, which featured 1-bit based players, but were quietly walking back the principle superiority of SACD from being 1-bit to being, well, whatever the marketing agency could think of.

But why all this fuss and bother if the first SACD players sounded just fine?  Well perhaps it's not just about making things good, it's about making them better and better year after year.  It was too hard to keep making 1-bit better (through various trickery and overengineering?).

Anyway let me also say that DSD is perhaps not as bad as I've written about here-to-fore.  While a pure 2.8Mhz 1-bit delta sigma system would be horribly information-lossy, and correspondingly noisy, that isn't at all like what DSD is.  DSD achieves low audible noise, AND low information inefficiency, by noise shaping.  Noise shaping is not just an add on, it is fundamentally what makes SACD possible, and that may be what I was not thinking about properly.  It's improper perhaps to call DSD a Delta Sigma system, it's a Delta Sigma Noiseshaping system.  Noiseshaping takes the place of the structuring imposed in PCM by coding.  That's the difference here: noiseshaping vs coding.

Noiseshaping compensates for the information loss below 20kHz in a comparable 1-bit delta sigma system, and somewhat more.  If it weren't for noiseshaping, the noise would be the clear sign of information loss.  But actually DSD achives better noise performance than CD in the midband.  Therefore, it is transmitting more information there.

And here, I'm not sure how to make the full comparison.  One way would be to consider the full band 20-20k noise performance as showing the "information loss" of the system.  And the other way would be to consider something like the A-weighted noise performance as showing "the effective information loss" of the system.  By the latter measure for sure, and I'm uncertain for the former, DSD arguably achieves not information loss compared to 16bit 44.1kHz PCM, but information increase.

There's no question still that DSD is an inefficient system in transmitting information, PCM is far more efficient.  But it could be said to be satisfactory in total information transmission, having reached CD quality and somewhat besting it, and having different character in the details, in ways that could be audibly pleasing.

Anyway, I'm thinking that the noise performance is a good indication of information performance, and one can just look at the noise curve of DSD and say it has more information than 44.1/16 in the midband.  I'm only not sure to which the increasing noise in the upper octave cancels that anymore, it might not cancel all of the advantage.  But anyway, the noiseshaping component means I can't simply apply my previous simplistic calculations.  I have to account for the effect of the noise shaping in shifting available information bandwidth from super high frequencies down to useable ones (in the reverse direction to the noise).  So another term might be information shifting.

And I wouldn't worry about the gaussian noise as much as tonal or correlated noise, and noise whose proporition increases or changes character at lower levels.

One way to explore "the details" in the sound would be to make a PCM or DSD recording at artificially low recorded level, say -60dB.  Then playback with lots of gain, and see which sounds better.  (This is actually more complicated than it sounds.  Should the gain used be PCM-like or analog, for example.)

Of course we don't really have to be cynical to see that the big win for Sony with SACD/DSD would have been IP and the big industry selling point was DRM.  Meanwhile I don't doubt that the lack of openness was a big hinderance in the end.

I don't think it's possible to consumers to make SACD's the way they could make CD-R's and CD-RW's, and that's part of the plan.  Interesting that the earliest SACD machines from Sony may not even have supported CD-R and/or CD-RW, though most CD machines of the time did.

The lack of CD-R/RW capability may have been a subtle hint to industry.  We own this, and we're not going to let consumers destroy your profit margins by permitting consumers to copy.

Anyway, WRT Sony, they visibly dropped any major concern for SACD and DSD sometime around 2006 when the Blu Ray vs DVD HD war was heating up.  SACD had not gone the right way, they might have figured, but lessons had been learned to win the next corporate battle, which Sony did.

Saturday, September 17, 2016

Legato Link

Legato Link is Pioneer's trademarked oversampling.  In constrast to most oversampling with brick wall filtering, Legato Link deliberately seeks to capture high frequency information above the Nyquist Frequency (?).  Some say it's merely a relatively slow rather than sharp brick wall digital filter--which might roll off highs earlier (but no evidence of that) or let some aliases through, and that appears to be possibly true, distortion levels would appear inconsequential to specification readers (the PD-75 for example specifies 0.0018, which would be -95dB, very respectible, out of -98.08 perfection).

Pioneer was doing interesting work in digital filtering from the beginning.  And so were others.  Pioneer used the term Legato Linear to refer to the highly refined filters in the PD-95 and comparable transport and recorder.  In my mere PD-75, they used no such term, however as it was the A version and the PD-95 got the B version one suspects it was already developed along a similar direction.  What Legato Linear means is not well explained, though we can guess the Linear has to do with things like linear phase response and/or linear filtering.  There's little evidence that Pioneer was attempting to reconstruct high frequencies, as claimed for Legato Link, though the distortion seems to be slightly higher than absolutely necessary, 0.0018% is, so one expects a compromise is being made with that and pulse accuracy, and if that's the case then only 0.0018% is doing very well, say compared with those now who would go to 1% and beyond with NOS.

The actual usage of the term Legato Link and the hype about reconstructing high frequencies only occurrs much later, possibly with the CXD-500, and then becomes commonplace (with additional adjectives, like Super) in the Pioneer Elite DVD players (and, sadly perhaps, one reason I never bought one, I didn't trust them, as I should have).

However, it appears by some measurements, that actual performance fell a bit short of those inconsequential looking specs, still somewhat insequential looking:

Peter Aczel, for example measure -92dB above 800Hz, but down to -78dB in one channel, or 0.012%, around 250 Hz and below.  He got identical performance in two units.  Published in The Audio Critic.

Julian Hirsch got 0.004%, or -88dB, 20-20kHz, which he seemed to think was quite good.

(These of course inconsequential compared with NOS designs, which may get 10% distortion in some audible frequency ranges.)

One interesting question here, is this digital distortion, which might be much worse as it would increase proportionately at lower amplitudes?

Perhaps the use of Spline interpolation prevents the digital distortion from increasing so much at low frequencies.  Anyway, now I'm a bit curious, and not at all turned off as I was by a friend in 1991 who showed me the pioneer advertisement:

I remember being shown this advert flyer during the early 2000's and I thought it looked pretty stupid.  I didn't then think that the high information retained would be anything but noise, aliases, and distortion.

I looked at that and said, that looks like aliasing distortion they're including.  They can't include high frequency information.  (Maybe they make it up by reasonable interpolation?)  So it must be aliasing and grundge.  I was highly skeptical, and moreso after reading various comments, reviews, and such regarding Pioneer machines.  I mean, I wasn't into buying machines like that much anyway, I was satisfied with the performance of my Sony 507ES I purchased in 1989 and still works today (thanks to a bit of ingenuity on my part, partly).  Anyway, Legato Link seemed like some sort of dangerous nonsense.

Meanwhile, in a somewhat pricier machine, also using bitstream dacs, Sony was getting -96dB 20-20k, and -97dB midband.  (That was for the X997ES, one of the best measuring machines ever, since then it's been downhill mostly for CD performance.  Perfection, again is -98.08dB and 0.001247% for THD+N.)

So color me as believing distortion perfection was being sacrificed in some way for time perfection.  Ok, I'm much more interested now than I was in 1991.  Very interested indeed, a lot of things from HDCD to SACD to HiRezPcm to MQA are all (or partly) about improving time accuracy, such as transient response.

For that reason and others I have some interest in the sound of the PD-75 with Legato Link and Pioneer Bitstream DAC's (which, btw have essentially perfect linearity down to anything you want, and have shown in some comparative measurements to have been, generally speaking models of comparative perfection) even if I've been (and will continue to be) doubtful of the full fidelity of anything 1-bit.

I really got the PD-75 as a transport, because the big mechanism, Me Tarzan or something, now it appears the clock is all important, wimpy transport with good clock is fine.

Still may be a decent transport, esp., with upgraded clock.  But perhaps just use Onkyo RDV-1 already having fairly decent clock, and (for sonics) 1704's.  And tis transport for anything allowing digital output.  So this may be my goto-transport for now, actually for the best sound, until and if I keep (probably) and modify the Pioneer (I may just play the Pioneer as transport, as it works, for it's "vintage" sound, jitter and all, probably still at low levels, maybe even unmeasurable contribution to jitter from the transport itself, just enjoy the vintage feel, and sometimes even the perhaps slightly amplitude distorted sound--but with better time response.)

I have an Oppo BDP-95 available through about 100 ft of high end coax wire (my home wiring network plus interconnects).  I'm not sure how much the spdif degrades, but I get perfect connectivity through my old Tact at 96/24.  (88kHz on the Tact died even with short cables long ago, but 96kHz works fine)

SACD and DSD in DVP-9000ES vs HDCD

I take this mostly as representative as SACD the system, not my particular unit or the DVP-9000ES more specivically.  Because it's how I'm thinking of it all.  I think this unit represents DSD as it was first implemented, with something very much like the 1-bit converters Sony had been using for years, and not the oversampled multibit implementations, which have been the norm ever since the 3rd generation or so.  And it's characteristic of DSD itself, in any implementation, but more purely in its original implementation, where the limitations are more obvious.  So I'm going to say DSD and not "my DVP-9000ES" but you can do the translation back if you don't accept that equivalence.

DSD has a smoothing effect mostly.  That's actually the subjective effect of adding noise above 10kHz, and possibly that's the cause of it.  But it can also open up dynamically, and suddenly get more bite than PCM.

I've recently noticed HDCD has an effect like that, apparently under the producer's control.  The producer can apparently control the sharpness of the digital filter in your system.  Some make it sharp, some make it soft, RR recordings you never notice because they do it frame by frame with the PMI recorder.  I think this is a good idea, to give the producer some control over the playback digital filter, and I like HDCD, perhaps an open standard system would be better than a proprietary one, but HDCD recordings as they exist are generally quite good and advantaged slightly over PCM 44.1/16 as a result.  There actually is a tiny information loss to power the coding system, but it's negligible compared with DSD.  Only a few bits are necessarily stolen by HDCD in full decoding (of course in non-decoding you are possibly getting a compressed form, so that part was not helpful to gaining wide acceptance--the "compatiblity" claim is actually overstated...then again the story in standard CD masterings being all over the map and endless remasterings to endlessly bilk consumers is no pretty picture either...there's really no compatibility in CD's every version is different, and one might be suspicious in hybrid CD/SACD discs that the CD is somewhat dumbed down to make the SACD shine, and in fact if they use 1-bit conversion throughout that would probably be the result).

DSD has this opening up and closing down aspect...but it appears seamlessly as part of the system, apparently driven by the demand for it in the music.  So that's good in some ways, less possibility of error, but I don't think it's much if any better than 44.1/16 PCM and I'd rather have HDCD or High Rez...but perhaps the same actual mastering used in the high rez recordings would work at 44.1/16 too, if they'd just do that.  Mostly the whole "High Rez" thing is to force people who want better masterings to pay more for everything.  It's a bribe to the Engineer not to crank up the compression machine.

I still feel the information loss in SACD (which I admit, is equivalent to the HF noise above 10kHz*, so well you might say, why worry, but there it is, that's what information loss looks like) gives the sound a simplified character generally, like MP3,  only very very slightly, just enough to make things sound better a bit, like blur on a photo.  Generally.

(*And I'm not exactly sure how to count the noise shaping, it hides the information loss within the audible range by creating vastly greater information loss above, but it can't add lost information.  Information loss need be applied to the raw S/N curve, not the noise-shaped one.  Actually, in a wide band, that would probably make it seem better anyway, but my information loss argument applies strictly speaking to the known audible range, and in that case I think it's pretty big.  I'm not exactly sure how to square this with apparent low noise in the midrange in measurements of DSD...but once again I think that's primary the bogus result of noise shaping, without which there would be noticeably more noise in the midrange and upper midrange.  In principle, despite noise shaping I think, DSD adds information up to the actual nyquist frequency, say 1.4Mhz more than equal to it's loss within the audible range, it uses lotsa bits.  But who cares about added information above 100kHz, which is filtered away in the end anyway?  Am I missing something here?)

So I wouldn't recommend DSD, but if a producer has chosen it, it's just another artistic brush, may not kill things for the first few hundred listenings, there's really so much information in music anyway, most of the time we're throwing away almost all of it, only sometimes reaching more than a tiny fraction.

Friday, September 16, 2016

Early SACD player reliability

After I first set up my newest acquisition, a Sony DVP-9000ES with new laser (and one spare), it wouldn't read the first hybrid multichannel SACD I put in it.  But after playing a Sony SACD for 15 minutes, I tried the hybrid again and it worked.  This may have been a warmup/settling issue.  I hope I've been charmed to see the first time fail, and every time after succeed...

This player is indeed full of calibration features and I think it may also autocalibrate on every disc.  Despite all that, it appears now that these Vintage SACD players may have reliability problems now.  One longtime modifier says that new lasers only seem to last about 60 days anymore, and he advises people NOT to buy this unit.  Some think it's because of the lack of quality in the current replacement lasers now that Sony no longer makes or sells replacement lasers for this unit.  It could also be some other kind of internal deterioration.

But, I remember the early days of SACD, when I was reading about the first SACD players like the SCD-1 in the pages of Stereophile and The Absolute Sound.  And I remember reading about things like "sudden laser death syndrome."  It seemed like many of these early SACD players would not make it through their warranty period without at least one factory authorized laser change.

I think somewhere along the line I read one commentator, was it HP, who recommended powering the player down after the end of a listening session.  All to hopefully preserve the life of the laser.  (I wonder if players cut off the laser when no disc is playing, or keep it running whenever the player is on.)  If failure corresponds to the number of hours left on, and 60 days being left on all the time is the maximum, I could get as many as 2000 disc plays before failure.  THAT would probably last me 3-5 years, especially as I will only use this player for SACD's which are a tiny fraction of my disc collection now.

I remember reading about all the earliest Sony and Philips SACD players of having premature failures.  Given the history of Philips in making solid stuff, like the CDM-1 transport used in most early Philips players, one would think they wouldn't have reliability issues, but judging by stories in the magazines the SACD-1000 made by Philips was the least reliable player of all.

Here's another discussion of issues with DVP-9000ES, but started by a guy who finds it amazingly good sounding, better sounding than his later and more expensive SCD-XA777ES.  Once again a description of how to enter service menu and do calibrations and check hours.

And, there's really no excuse for not keeping backup parts for at least 10 years, and I think Signature products (like ES or Elite) should have parts available for 20 years.  (Pioneer may have been somewhat better, I think PWY1004 laser required for PD-75/91/93/95 players was still available until recently, or maybe even still, and those are 25 yo players.)

Sony DVP-9000ES owners were complaining about lack of parts in 2006, only 6 years after player was purchased.  They're saying ES stands for Execreable Service.  Not mentioned there, a cache of 9000ES players was discovered in Mexico in 2005 and sold through a major non-dealer, supposedly with original 5 year warranty (I find the latter hard to believe, I think these were Grey Market with any warranty provided for the dealer--for a day or two while the blowout seller still had old stock on hand).  The discussion ends with praise for Oppo who apparently fixed a guy's player long out of warranty--for free.

It does really seem that in the 2000-2010 era, Sony burned their bridges with a lot of people by selling megabuck disc products they failed to back up with good parts and service.

My 9000ES arrived in a pristine factory box and was packed like it just came from the factory, as did the box.  It does indeed look mint as the ad said.  This seller apparently babied this equipment as Japanese do.  But now I'm beginning to wonder how good a deal this was if I have to replace laser every 2 months.  In this case, I'm keeping the box incase I decide to resell after I've had enough.

Here's one endless list of problems with SACD players of all brands.

Now I think I may remember one trick.  Mostly don't leave a disc in the player when powering off.  This engages the auto-calibrate mechanism when the machine is started.  This wastes laser life.  However, If having trouble with a particular kind of disc, you deliberately leave it in the machine, and power cycle.


OK, I've played a lot of discs, then with the Mercury Living Presence Hanson, the 9000ES stopped with C 13 00, which apparently means "dirty disc" (the usual attempt to blame the user for the machines limitations) or, perhaps more correctly, disc read failure.  On a later attempt it had skipped 3 times before reaching that stage, perhaps it had skipped the first time also (I wasn't paying close attention).

So, I think it was precisely regarding Mercury Living Presence releases that I read the trials and tribulations of early SACD players (for those guys, probably mostly SCD 777ES).  In fact it might have been precisely this Hanson disc.

Anway, I found this thread about C 13 00 appearing in 9000ES (and many other Sony players).

Well they they do talk about service menu calibration, but they still don't talk about the simplest form of recalibration, simply leaving disc in machine and power cycling.  I think what this does is re-register the disc.  So it identifies the disc type rather than reading from memory (which apparently it also does) so maybe it doesn't actually do any recalibration, but it might I think.

Anyway, after I first got the error, I tried play again, and now it recognized this disc as a CD.  I put ejected and reloaded and it still recognized it as CD.

So here I put the disc in and cycled the power.  And sure enough, it now recognized the disc as a SACD again.  So I played.  And it failed at what sounded like the same point, second track.

So I played an SACD from Sonoma, also multichannel, which played flawlessly.  Then back to the Mercury Hanson, and, it failed again at the same point.

Later SACD players didn't have these problems.  I don't remember a single read error with my $149 Sony NS500V.  Guess I'll have to keep that, and it also makes me wonder about the possibility of Frankenstein.  Because I do love the sound of the 9000ES better than anything I've heard playing SACD, if it would only play all of them.

On the plus side of reliability, Stephen Sank says he's never seen the laser go bad in a PD-75/91/93/95.  And he's worked on hundreds.  Sure the lens falls out, but he's figured out the intricate procedure of mounting it back on correctly, getting all the correct technical patterns and so on.  Which BTW in the service manual is very very good.  Pioneer has pages of oscilloscope photos accompanying their adjustments.  From what I recall of Sony machines, there may not be so many adjustments, maybe just one, and if that one doesn't fix it, it's cooked.  Now I've read David Rich say the fewer adjustments the better.  But this may not be true in all cases.  I think in the case of something like PD-75 being able to fine tune the system in all different ways is excellent, and prevents having unservicable machines down the line, when situations arise that the onboard computer was not designed to correct.

But those were slow speed CD players.  Once we got into the real of high speed discs like DVD and especially it seems SACD there were a lot of early failures in the first few years, then by about 2004 or so machines could be made which last till today essentially like new, such as the Denon 5910.

Denon btw does seem to create something more like heirloom products generally.  But even they had problems in the early days of SACD, I suspect the laser in my 5900 was replaced by the previous owner.  I have two inoperable 5900's.  And of course my DVD-9000 no longer plays DVD's, only CD, a common problem with DVD-9000.

Tuesday, September 13, 2016

Disc Players and Transports

I started using a network audio system in 2006, Sonos, and I still believe it is a fine system fundamentally.  I Sonos Connect nodes (originally called Zoneplayers) to provide coax SPDIF digital signals to systems in every room, mostly from my kitchen Mac's hard drive (a 1Tb Fusion drive), sometimes making connections from analog sources in different rooms.  All digital signals are uncompressed 44.1/16, which I've always believed to be fine and good.  John Atkinson measured remarkably low jitter from a Sonos based wireless system in 2006.  I actually eschew wireless and hard wire all my nodes with Cat-6a ethernet which permits attaching analog sources in 5 different rooms simultaneously for uncompressed forwarding.  All nodes are individually wired to a central gigabit switch.  I've often dreamed of DIY replacement for this, but it's almost unimaginable with my time constraints now.  Newer network audio systems Never (as far as I know) permit attaching analog sources in different rooms.  Sonos was the first and only to offer analog inputs and it was why I chose them instead of Squeezebox, though I missed the Squeezebox high rez capability.  And most new audiophile computer audio systems are astronomically expensive, like $20,000.  I bought Sonos Connect nodes like candy because they're only $395, small price for another room of fully 2-way connected audio.

It's very convenient, but has never played high resolution sources, so I maintained a collection of disc players to play high rez such as DVD-Audio discs (which are sometimes incredibly good), DAD/DVD-Video discs by Classic Audio, which are all fabulous, and SACD's, which I consider pop high rez (widely used if not the best system) but still indispensable.  And HDCD's too, which are all among the best recordings I have, and the more I know about it the more I still think HDCD wasn't a bad idea.

Are Disc Transports better than Computer-network systems?

Now I've even had my mind turned to the idea that disc players may be different and possibly better than computer network based systems.  In any case, it's an excuse to play ordinary CD's on my Denon DVD-9000, which is just about as fun as clicking on things on a computer screen, plus it has class (and weight!).  In most cases now I take the analog outputs from disc players and resample them to digital (for my crossovers and DSP) using a Lavry AD10.  So, FWIW, I am preserving the sound of the different players, to the degree that it can be done this way (and it is my belief that the 96kHz/24bit sampling of the Lavry is extremely good and transparent enough to do this).  I haven't seriously explored the "better sound" I get from playing discs as compared to network bits, but in a few cases I've noted the difference between taking the analog outputs and the digital outputs, and in almost every case I've preferred taking the analog.  I consider my Lavry resampling to be as good as the best preamps, and probably better than most preamps.  Digital sampling sounds impossible but has been incredibly well developed by our technological society.

As someone with collector tendency, however, somehow I wouldn't mind having EVERY classic disc player in the rack, so I could try them all and vary according to mood.  Unfortunately my latest rack was only just set up a month ago and already out of space.

Anyway, I've long felt I wanted a Real SACD player, and not a Universal not particularly noted as having the best sound on SACD's like my Denon DVD-5900.

I've long scanned eBay for SCD-1's and SCD-XA777ES's, the two original and very similar super high end SACD players by Sony.  To me, these represent REAL SACD, as Sony originally intended it.  I don't necessarily consider these players to be any sort of ultimate for regular CD's, after all Sony was primarily hoping to sell the idea that SACD's were superior to CD's, so one would imagine these players pulling out all the stops for SACD but possibly falling a little short on CD (and some reviews bear this out).  Some do say some later machines are actually better on SACD also, such as the much more cheaply built but still pricey XA-5400ES.  Somehow that latter machine has never appealed to me much.  Anyway, even if it did, I wouldn't want to spend that much as current used pricing for such a lightweight machine.  (I could be tempted to spend that much for a heavyweight universal, such as an Esoteric UX-3.)

I especially loved the balanced outputs of the SCD-1, which would be fabulous into my Lavry.  I also noted the especially clean noise floor of the SCD-1 in Stereophile review, and I attributed that in part to the balanced outputs (and multiple overbuilt power supplies a factor also).

But both those players had a very fancy but notoriously unreliable CD mechanism, the "fixed laser" mechanism.   There doesn't seem to be any real advantage in making the CD spindle move instead of moving the laser.  It just seems to be something rather difficult to accomplish, something that few companies would even attempt.  Sony of course did claim it was better, but there are many machines people believe to be better transports than the SCD-1.  To many observers, and to me, the "fixed laser" motor seems to be a marketing gimmick with no real advantages, and something that saddles the owner of these players with future repair costs.  Sony never seems to care about that, in fact their history has been one of a very high level of planned obsolescence, including not making stuff to last very well beyond the warranty period.  This is probably why Sony, which now seems to own just about everything, should be the most successful company ever but instead is barely keeping the lights on in their vast empire.  What has happened is that generation after generation of new customers has left Sony behind after being burned one time too many.  I've been among those burned many times, though I'd still been a big Sony customer until very recently (when they don't seem to make anything I want anymore).

Anyway, the 3rd Sony SACD player to have SACD electronics reportedly similar to the SCD-1 (said to be something like 75% the same) is the Sony DVP-9000ES.  There is hardly a time on eBay anymore when there aren't a half dozen or more of those for sale.  After years of scanning for them, I've gone ahead and purchased one with a newly replaced laser tested on multichannel SACD's--the hardest to play.  This cost more than the current average price but I hope will be worth it.

The DVP-9000ES has a specially enclosed but ultimately ordinary DVD/SACD/CD mechanism as far as I can tell.  The laser is shared with other units and plentifully available.  I don't even know if it's all that special but people have said the 9000ES is a decent transport...perhaps as good as the SCD-1.

This machine will be my go-to machine for SACD's.  For CD's I'll stick to the Denon DVD-9000, the differential 1704 based behemoth.  For DVD-Audio and DVD-Video-Audio I'll use the Denon DVD-9000, my current Oppo BDP-95, or perhaps get a new BDP-105 for the living room.

I would have preferred to do DVD-Audio on my DVD-9000, but that feature is broken on my unit now.  I could also repurpose my Onkyo RDV-1 as a DVD-Audio player and use something else, perhaps my Denon DVD-5000, as a supertweeter DAC.

Does a transport make any difference?  Well I don't know, but I imagine if the transport is temporally unstable to any degree, this is going to affect everything downstream to one degree or another.  I suppose instability could be erased simply by reading an entire song into memory.  FWIW, people I know had disliked memory-based players.  At the risk of sounding technically illiterate, I actually don't know if any electronic systems have less jitter.  I've very suspicious of all asynchrous digital systems like ethernet, and I have always thought USB to be subject to the same highly jittery phenomena but it might have a kind of "synchronous" mode, I don't understand that but I still think it's not fully synchronous.

One has to understand that even if things are buffered, reclocked, or whatever, at the end, those subsequent phases will ALWAYS have to adapt to the the source matters.   (And a jittery transport layer matters also, though entire-song delivery would seem to be safe, so where is it unsafe, I don't know.)  It may be in principle that something based on a later clock can simply clock things out.  But if _it_ has to wait longer or shorter, that may cause some unintended small difference, say from the heat in the circuitry for having to wait longer, or negotiate with the source for more data packets more often.  So one can try to wall off noise in the time domain, but no such wall is ever perfect, computers don't really work on 1's and 0's, they work with electrons and semiconductor junctions, everything shifts a little, and for most work the absolute clock stability isn't important once you're off the chip and even perhaps there.

So I really don't know if semiconductor memory in a dedicated circuit, or any actual circuits, would abolish jitter any better than a spinning disc.  I think it probably would, but I don't actually know.  It could well be that if disc transports sound different than electronic sources, it's because the disc transports are expressing their character (of some kind) in the timing.  And in this case, where it isn't captured by conventional measurements or ideas, I'm all for this kind of difference.  I want to try all the different flavors of candy in the shop of my imagination!

So what looks good to me?  Well, the VRDS Neo players from Esoteric, I'd love to have UX-1 or UX-3 because those are Universal Disc players, and that's ultimately what I need.  Even if it isn't Sony, or even "real" DSD, it's close enough to be a Universal for me,  high end mechanism and quad 1704's.  These are the most heavyweight low resonance mechanism ever.

I'd love to have a CEC made belt transport, even in the Parasound 2000 version (which was reputedly based on CEC's cheapest model).  You can't argue with top load and screw down, that's always been the best way IMO.

Pioneer's Stable Platter machines look interesting, but only a few use "the real" Stable Platter with heavy parts.  These were incredibly high end machines I knew nothing about since I wasn't much following high end audio in the mid 1990's.

Sony did make a few excellent pieces of machinery in the 90's.  I believe the XA5ES, one of Lampizator's favorites, is one of these.  After that, and after the whiz bang SCD-1 and SCD-777ES, it's been pretty much all plastic.  I think the key assembly of the mechanism of my 9000ES is all plastic, if not the entire thing.  I recall my 507 is largely plastic.  (I fixed the 507 mechanism replacing the quickly broken rubber band with a weight.)

Phillips really did more in the direction of engineering things to last (but then, sometimes didn't).  Famously they made the CDM-1 mechanism and a large number of all metal derivatives.

The original Theta Data transport was based on a Phillips Laserdisc transport (basically, that's all it was, with thick panel added and tiny proprietary board added, which according to Lampizator, doesn't do anything useful, but he could be wrong, and price raised from $500 to $5000--US Engineering!).

Nevertheless, Lampizator has loved that transport and thinks there is magic in the much heavier mechanism and 30 times more powerful motor in a laser disc player.  So....there's a wide field of interesting looking laserdisc players.  Notably the top Pioneer players, that look to have excellent mechanism and digital output, but more research necessary.  Meanwhile I find the top Pioneer Stable Platter players, a cheaper version of which was used in a Wadia player, most interesting.  And I had not known at all about such things until now.

I do think that keeping the disc stable and clamped to a heavy disc on one side and tight clamp on the other is a most excellent idea.  I have long liked serious looking players with user tightened weights.  The SCD-1 appealed to me but now I see was not the giant I thought, but I still like the top load idea, Krell did some most excellent top load CD players and there may have been others. But it seems Pioneer was accomplishing this full disc clamping idea through a well engineered mechanism long ago, with just an ordinary (well, large) drawer.

For those with transport obsession (or is it transport lust?)  one essential guide is a transport list, this is a good one, though I think I've seen others that are about the same in comprehensiveness, and nothing is ever truly complete.

Every time I look at these lists I find interesting factoids.  For example, my Denon DVD-9000 (aka DVD-A1) uses a Hitachi HOP-1000.  That rings true as I know it needs a Hitachi laser.  The Esoteric X-01 and X-03 use variants of the VRDS-Neo and the UX models aren't even listed (but I think they use the corresponding transports).   The Onkyo RDV-1 uses a Mitsumi PVR-202T.  And, what I was just checking for today, the Parasound C/DP-1000, C/DP-2000, and C/BD-2000 (transport only) all use the same CEC transport, the CEC-Sanyo SF-P1 / SF-90.  So guess which model is the easiest path to belt drive CD nirvana?  In the Pioneers, the PD-75, PD-77, and PD-95 all use the ultimate "magnetic stable platter" PWY1004.  CDM-1's can be obtained in a number of Proceed models.  My old Sony 507ESD uses a KSS-151A.   The later Sony 990 favored by a friend sometimes used a much cheaper looking KSS-240A, which is among the most widely used transports ever.  The expensive but problematic SCD-1 mechanism is the KHS-180A.  Teac (the parent company of Esoteric) makes some models that claim to have a VRDS mechanism like the big dog Esoteric's.  But they don't, they have the VRDS CMK which looks like a barely warmed over Sony mechanism, which it appears to be, as the transport also includes the suffix KSS-151A--the same as my respectable but ancient CDP-507ESD.  Interestingly enough, a number of well known Wadia models use the VRDS CMK KSS-151A transport also.  BTW it was a broken spring, not a broken rubber band, that I replaced with a weight.  The original spring had gotten too weak for the mechanism door to work.

Here's a discussion of the various top Pioneer CD players from the 1990's with Stable Platter.  The biggest of the big dogs is PD-93, followed by PD-95, PD-S95, and PD-75 which all use the same Magnetic Stable Platter mechanism I listed above.  And also the PD-91, PD-73, and PD-77.  Other lesser Stable Platter machines include the PD-65, PD-54, PD-52.  The Burr Brown PCM63K equipped models are the PD-93, the PD-73.  Much as I enjoyed the discussion, it's critical to check back on the mechanism list to be sure what is what.

Question: Do transports even make a difference?  Well if they don't, or if even if they do and systems based on USB or Ethernet are better, we could just kiss transports goodbye, except for the "fun" of putting discs in them, and for that we could just get the cheapest available unit (not sure of how many there are anymore, but tons of old classics on eBay).  Except I personally get more satisfaction from putting my CD's in a big heavy expensive and rare player, especially if it was widely regarded as being the best.

For most of the time since CD was introduced, I haven't believed that transports make any difference, and I still don't necessarily believe they do.

As long as the bits are being decoded properly (and it is usually assumed they are, but I think periodic maintenance testing may be called for) the only difference the transport could make would be in the timing.  Rather than emitting the digital samples at a perfectly stable rate of exactly 44.1kHz, it might vary slightly.  This is called Jitter.

And this can (in principle) be measured directly or indirectly (it is usually measured indirectly by looking at the distortion spectrum), but few people do these measurements and publish them except for John Atkinson of Stereophile.

The issue that separates the objectivists from the subjectivists is not whether or not something like jitter is possible, it certainly exists!  The issue that separates the two camps is whether jitter in actual digital systems is audible.

The best and just about the only objective research on that topic was done quite awhile ago and published in JAES.  Basically they found that jitter because audible at levels about 10-100nsec, which is about 100-1000 times greater than is found in typical digital audio systems (say, 250psec).

So that's it, and I had pretty much accepted that view for the last 15 or so years, and even before that I highly doubted the audibility of jitter.

While I strongly doubt that there is this kind of difference between players, I still want the best one with the lowest jitter and the most substantial build quality and design quality, etc., just because I deserve it just as much as people who think, wrongly, that they need it.

Anyway, here's Lampizator's take on the audible differences.

I have this different theory that like looking at nice stuff, and then we enjoy the music more.  But anyway, here's another discussion on the transport audibility.  Many in this group are using a what is a pedestrian DVD player now (the Denon DVD-2900, which may have originally sold for $1000 in 2002 but now typically sells for about $100 used) as their CD transport if not their CD player, and more focusing on getting good (or at least interesting) DAC's.  I would generally agree that the DAC is more important, but that brings up another thing--according to objectivists all DAC's should sound the same also, given distortion below 0.1%, except for NOS dacs which have distortion well above 1%.

My heart is warmed by seeing one guy in that group (and I know this is fairly popular) using a Sony DVP-9000ES as transport and/or CD player.  Because that's the machine I will be getting tomorrow.  To me, however, it doesn't look special either as CD player or as transport necessarily, I am buying it for the very serious 1st generation stereo SACD capability, said to be among the best.  As a transport, it looks to use a commodity laser and similar mechanism to that in other DVD players.

To me, a CD transport should be a dedicated CD transport and go beyond the call of duty in every way possible, such as holding the disc flat and not flopping and shaking.

I have seen what most people have not, the insides of a player being adjusted, and it's not a pretty picture, and it does even lead you to wonder how the CD system can work at all.

Of course the bottom line is stability, error free reading, and lowest of the low jitter, and also low hum and noise (because even on the SPDIF output, that contributes to jitter also).  By most reasoning, it doesn't matter how much metal the transport uses with the intent of reducing jitter, it's the actual jitter performance which matters, not the intent, expense, effort, or intelligence that went in to reducing jitter.  The problem is that it's not so easy to measure jitter and errors, and it can't be done all the time until we retire the player many years later (if it's a good one, and possibly never if possible).  So we look to surrogates of good performance, like "build quality", or fancy mechanisms which clamp discs tightly or are claimed to reduce vibration in some way (which could all be marketing hype, as it could have been for the fixed laser transports).  The more effort and expense was put into making something better, the better we think it probably is.

So I laugh with Lampizator when he laughs at the real deal VRDS-NEO transports..."lots of heavy metal, if you call that engineering."  But I also laugh at him, and me, for dissing all the cheap lightweight plastic transports anyway.  Of course it should be understood that most intensive engineering actually goes into designing the plastic mechanisms.  They take lots of engineering to do correctly, including perhaps such things as finite element analysis.  With big heavy slabs of metal, you can just pile it on to your heart's and wallet's content.  And if Esoteric is doing the piling for you, it's going to cost many kilo bucks.

(Lampizator is very inconsistent, BTW.   On the one hand he liked the Theta Data transport because it's a big Laserdisc transport with big motor.  Elsewhere he says smaller motors are better for good sound.  And so on.  He's unbelievably consistent in another way.  Everything he modifies becomes the best thing ever.  But he's fun to read and especially the dissections of CD players is interesting.)

Lampizator did an interesting dissection of the Audio Alchemy DDS Pro.  According to him it is basically just a Pioneer PDS-502.  And he think compared to standard mechanism, the cheap version of Stable Platter is all plastic, every significant part is plastic, including spindle and bearing and it wobbles like hell, and though it works fine, ordinary middle of the road Sony and Philips transports of the time were high end by comparison.  This makes me feel good I got the PD-75 with the largely metal Magnetic Stable Platter which has features like a heavy metal platter and spindle, sapphire bearing, and a motor with far more torque than most players.  Below, however, Stephen Sank has a different view, he recommends the Audio Alchemy to others even though he has one of the big dogs.

BTW, regarding the Denons, the transports are generally metal.  The HOP 1000 mechanism in my DVD-9000 is all metal, looks solid and ok, but not overbuilt though, the cast metal early Philips CDM mechanisms looked far stronger (from pictures).  And there isn't necessarily anything special wrt how the CD is gripped.  I think that detail is too often overlooked, which is why I like the Pioneer stable platter idea.  Even top loading CD mechanisms that require you to hold down the disc with a magnetic weight still have essentially let the CD flap in the breeze.

Did Pioneer lose the 1990's CD wars because Sony had better sound, as The Vintage claims, or better marketing.  I think I'm more cynical.

A guy at DIYAudio did distortion spectra testing with a number of CD players and DAC's.  Guess what, one of the best of the bunch was the PD-75.   TDA 1541A looks pretty awful actually, and TDA1540 one of the very worst.  PS1 looks horrible.  1702 is not good at all, 1704 better but not perfect.  His favorite chip is a particular PCM56 he found, which adjusted looks just about perfect, but he notes that other PCM56's have been far worse.  PCM63's in PD-8500 looks awful.

Strangely however, people find the bitstream chips in the PD-75 to sound particularly undynamic, and the TDA1541A is more popular than ever.

It really looks as though people prefer distortion.  But perhaps the guy above didn't do comprehensive enough measurements.  It turns out The Audio Critic did a review of the PD-75 when it was introduced and found very clean highs but excess distortion (still at levels believed inaudible) in the bass.  That does suggest something wrong with the audio section on this unit, but not likely the transport, as if the transport were adding excess jitter that would be as much or more evident in the highs.

In that review, the reviewer said he saw no advantage in putting the disc upright (or clamping it onto a felt platter, though he didn't mention that) except that less dirt would fall on the laser.

Unfortunately it appears that some time after its introduction, while much loved, the Stable Platter transport became infamous for dropping lasers onto CD's, or falling apart during shipping.  Famous equipment repair and modification authority Stephen Sank advises strongly against PD-75/91/93/95 for that reason and one other.  He himself uses the PDR-09, which features the same fantastic mechanism but with a much better laser.  PDR-09 is even more high priced and less attainable than PD-93.  For the rest of us pedestrians, he suggests the PD-S06, which has the more plasticky PD-65 mechanism but with a very good clock, and PCM1702's, very highly regarded for sound.  He says it makes a great transport, but you'll want to use it as a player because it sounds so good.

Well guess what there's a PDR-09 for sale on ebay right now which I could have budgeted for and might get someday.  The PD-S06 is widely available on European sites according to HiFiShark but not on US eBay.

Lampizator also likes the PD-S06, though he started out with his usual complaint about the supposedly-great "stable platter" mechanism being nothing but a plastic job like any cheap player.  But he doesn't inform us this is hardly the only "stable platter" the real stable platter machines are the Magnetic Stable Platter big dogs like PD-75 and PDR-09.  If he looked at one of those he wouldn't be able to make his joke.  So here we we see even in Lampizator as with all other reviewers that we never get the full story.

Is it All in the Clock?

Rethinking the ideal transport issue, it may be that the master clock within it may be the most important concern, as both Sank and Lampizator suggest.  Whatever the mechanism does, it ultimately feeds bits faster or slower into some kind of memory buffer.  The memory buffer drives the spindle motor servo to run faster or slower to keep the buffer at some middle level.  Only if there are errors or buffer under run is the mechanism an issue (according to some).  The clock ultimately clocks everything from the buffer into the internal DAC and SPDIF output.  So it's the clock, ultimately, which counts most (or entirely, according to some).

I argue that given a particular clock mounted to a particular machine, the drive also matters as the drive speeds up and slows down, changing the rate of information being added to the buffer.  As you get close to one end of the buffer or the other, I'm concerned about some kind of quantum time compressing effect which most engineers wouldn't be concerned about.  Also there is the unavoidable fact that different amounts of current will flow in the motor, possibly upsetting the clock.

So even in my far out view, the clock is still the most important, but if the clocks are otherwise equal, then the mechanism could still have some effect.

But how good are clocks to begin with?  I have read in fact that the designers of the CD standards already knew that the clocks were going to be very important, and they set standards, with two levels of excellence.  The first is 50 ppm, and the second is 1000ppm.  Presumably any high end CD player or transport would far exceed the first standard, and would that be audible?  I would think it would be already quite good enough, particularly with an "Elite" or "ES" player, good enough to not require a clock to have jitter far less than would be audible.

At this time I haven't figured out how to convert from these "ppm" specifications to the usual "picoseconds of jitter" specification.  And then how often do we actually know the jitter specifications, most manufactures don't give them and they are highly dependent on the equipment setup.  According to many, if your biggest concern is jitter your best approach is to buy the best possible "single box" player you can get, which only has analog outputs.  Then there won't be any issue with jitter caused by the transmission of digital signals outside the box, which is one of the biggest concerns in modern audio systems, at least in part because we use (even in AES digital) single wire (or differential, which is only slightly better) signals which combine the clock AND the data and therefore lead to the possibility of the highly undesirable data-influenced jitter.

One thing, unless I can determine that my PD-75 has far less jitter than anything else I have, I may not be happy with it until it has a sufficiently upgraded clock.  I'm realizing now that my Onkyo RDV-1 has a special Apogee sourced clock and maybe I should use that player as a transport (as well as using it for DVD-Audio…for which it has PCM 1704's).

In fact, I'd been thinking is that perhaps all a person needs is a memory player with a good clock.  This sounds like it would be cheap to make, but the only one I know of is the $4000 PS Audio Memoryplayer.

If it has a good clock, and the servo is strong enough to keep the internal buffer near midpoint, then all a good transport needs to do is provide data reliably.  But how do we know it's reliable?  Sadly few players have ever provided any way of knowing.  One of these was the Cambridge Audio CD3, which had a Disc Fault light (I would assume those show the hard errors that cause fill-in, not the soft errors that are fixed by CRC, we don't really need to know the soft errors, but I think what would be best is a counter that shows both types for the current disc being played).  The CD3 looks to be a nice unit with a Philips CDM1 Mk2 mechanism.  A transport version CD3M was also made and now priced many times higher at resale.  The companion DAC unit used 4 1541A's.

Measuring Jitter

When I started this blog, I had hoped to do objective measurements more frequently than, well, the sort of rambling I had ended up usually doing.  But it seems I have things I need to say even without doing measurements.

But I'd sure love to check out the jitter of different players and/or player configurations.  NwAvGuy, the guy who did excellent audio blogging for a few years then disappeared, gives the details here.

Other Info on Stable Platter players

This report gives far more details than the short review in The Audio Critic on the PD-75.  He measures THD as 0.004% from 2-20Hz, and 0.0035 at 1kHz.  That's excellent, and not worth worrying about, however it is as much as 9dB worse than the CD format is capable of (around 0.0015), so maybe that's in line with what The Audio Critic is reporting.  However reading this review about other design features and aspects of technical performance makes it sound like the PD-95 is one of the best players ever.  I'm not so sure about that, but it's comforting to know about all the good design features and aspects of performance.

Actually the above report may simply have been cribbed from the Stereo Review review of the PD-95 published in September 1991, as shown in this collection of Pioneer reviews.

The Audio Critic review of the PD-75 complained that the distortion rose to -78dB in one channel in the mid bass, and -83dB in the other.  What are these numbers expressed in percent?

-78dB is 0.0125% distortion about 4 times more than claimed by Stereo Review (they claimed 20-20k at 0.004%)

-83dB is 0.071% distortionn, about twice what Stereo Review claimed.

BTW, I think the best you can do with CD is THD -96dB, which would be:


though I'm not quite sure, I've also seen -98dB, which would be:


In any case I believe The Audio Critic when they say that the levels of distortion produced by PD-75 would not be audible, and they gave this player the "silver" award after noting in disappointment that the PD-73 had been better and winner of previous tests.  But the reviewer preferred to buy the Sony CDP-779ES.  He had to excuse himself, because there is no reason to buy such a pricey player (it was $1900) for audible performance alone, on that score a $20 CD player would be sufficient according to his endless preaching.  But he liked how the 779ES was made, and he said that for $12,000 he'd pass, but for $1900 he could afford and wanted the undeniable quality in excess of need.  I believe this was Aczel speaking, in a later set of reviews it was David Rich who bought the successor CDP-707ES, essentially the same as the 779ES.  Or vice versa.