Monday, March 31, 2014

More on audio objectivism

NwAvGuy has posted a great and lengthy thesis on what we can (and can't) hear.

I should read and digest this, especially now that since 1983 I've crept back in bed with the subjectivists.   I officially consider myself a "Grey Hat" who looks for plausible scientific explanations of what subjectivists talk about--beyond mere dismissal--while simultaneously saying that anything not proven in blind testing should not be considered "a big difference" as subjectivists are constantly doing.  My hope is to find things that have been overlooked in standard audio engineering, or maybe even further beyond, in information theory.  However I immediately concede that I have not done so, and that many subjectivist claims regarding how the objectivists were proven wrong are not accurate, and that overall the objectivists have been more right than wrong, and perhaps even entirely so.

Here's a similar thesis maintaining that 24/192 music downloads are very silly.

4580 IC's in Onkyo RDV-1

I was greatly relieved to find the Onkyo RDV-1 does indeed accept 96kHz inputs (in either coax or toslink, it turns out).  I run all my digital processors at 96kHz.  When I first opened the box I checked the manual, it said it only accepted up to 48kHz.  Despite my lack of time on Saturday, I quickly hooked up my new Black Lion Micro Sparrow ADC (mk1) to the Onkyo, and found it handled 96kHz fine and even shows a message saying it is doing 96kHz.  The Onkyo does not accept digital inputs higher than 96kHz, when given such inputs it displays the LCD rate, either 44.1 or 48khz.  I did not listen to see what those did.  But 96kHz in several tests over the weekend appears to be rock solid and sounds fine through the "stereo" outputs (not the same as the "front" outputs on this player…the "front" outputs sound very thin as subwoofer use is assumed--UPDATE, this may not be true with digital input--all were measured to be flat).  I measured the Stereo outputs with pink noise (which is random, so always fluctuating and never perfectly flat) and it looked as good as it gets with random noise measurements:

I've been pouring over the service manual schematic to see what the circuitry is like.  The output board has each output channel going through a 4580 opamp, then a 47uF capacitor, a small resistor, a number of shunt muting devices, and then the output.  Not wonderful but not too bad either.  I'd like to replace that electrolytic cap with a Teflon.  But what is a 4580 opamp?

Apparently it's similar to 5532, a higher current bipolar amp.  The 4580 is said to be much better than the 4558's used ubiquitously in "mid fi" audio equipment, maybe even some not so mid fi.

While researching this, I found the quotation from John Curl himself, as of 2009 his CD player uses 4558's (!!!)

Trevor, the 4558 is the IC that I have to listen through to compare SACD to DVD to CD. I can't afford an expensive player, because I don't design them.

I also found this objectophile blog saying essentially that 5532's are good enough.

I just checked photographs inside the Behringer 2496 DCX and guess what chips it uses?  4580's!  Somehow that still gets to about 0.1% distortion at full (just below 10v rms) output.  The problem with the DCX may be mainly that it insists on driving the outputs to 10v RMS at 0dB digital, and the only way you can attenuate that is by reducing the digital numbers, which is losing resolution and linearity.  And also that the DCX doesn't really have good enough power supply to drive the chips to 10vRMS with low distortion either.  To really do the 10V output well, it either needs better power supply, better chips, or both.

Meanwhile, my friend Tim is planning a significant chunk of work to add the best sounding of all IC's according to him and other sources, the OPA211, to our Kenwood L-1000T tuners.  That sounds good to me, so I won't be bugging him with links saying all good chips sound alike.  In fact I plan to address the psychology of those claims in a future column.  Tim analyzed a set of top opamps and decided which ones were actually most linear, the OPA211 on top, just as in the list he started with.  Here is the list he started with (not sure from where):


Compared with 4580's, Tim says OPA211 have 20dB more gain (which is good for opamp circuits, particularly ones with equalizing feedback) 80Mhz vs 12Mhz bandwidth for 4580, and distortion of .000015% vs .0005%--a factor of 30.  He says these specs (gain, bandwidth, THD) correspond generally to better sound quality (as determined in tests by him and a professor of engineering he collaborated with).

He generally doesn't respond when I talk about sighted tests, statistical significance, and the like, so I'm not going there.  It's not like even "white hat" audiophiles blind test everything to statistical significance.  The develop rules of thumb which extrapolate from blind tests, and then follow those rules of thumb.  But there's no telling whose extrapolation is actually better, without actual evidence.  All we really know from serious audio science is what the minimums are for good fidelity.

Thursday, March 27, 2014

How the SL 1000 was packaged

The SL1000 turntable base and attached motor (less platter) were packed in a makeshift oversize box which was made of two layers of corrugated box material salvaged from two other boxes.  As I opened the box from the top, or at least where the shipping label was, I first encountered 3 layers of bubble wrap covering the base and motor, with foam pads around the sides.  The turntable had been upside down, and looking down on the box I could see the bottom feet.  I carefully removed the turntable base and motor from the box, and after removing the bubble wrap, they appeared undamaged.

In several initial emails, starting from before I purchased the turntable, I told Mr. George Heid how I wanted the turntable pieces boxed.  He had agreed completely, and said he would box them as I had said or even better.  I also described how to do proper double boxing, with two inches of solid padding between the inter and outer boxes, and that the outer box should be a very strong box, preferably triple ply.  Even before I ordered the turntable I ordered a motor bracket to secure it.  I told him to wait for that, and he said he would start working on packing the turntable anyway.  It took almost 2 weeks for me to get the motor bracket, and then I sent it to him by 1-day express mail.  He then told me he would get the turntable packed and shipped that week.  At the beginning of the next week he said he was still working on it, but he would ship it by the end of the week, and so on.  Finally he did ship the turntable on the 45th day after purchase.  I only discovered around day 40 that I could not open a case through eBay buyer protection about non-delivery since more than two weeks had elapsed since the projected delivery date.  But he did finally ship a few days after I called the number he had given me earlier oat about day 25 in order to describe some better condition SP10 parts I could buy from a friend of his, especially a better condition dust cover (without the small crack shown in one corner) and asked me to make an offer (I made some low offers, still having confidence in him but worried about buying things off ebay, and also not quite appreciating the value of an SL1000 dustcover).

It was pretty clear from the moment I picked up the boxes at UPS that he had not done as I had said, and it was even clearer after I started opening the boxes.  So I was extremely relieved to see that the turntable base and motor were undamaged anyway.  After doing this, I left home for the workday.

I believed at the time that it was only the base and motor that were in that box, because there were three boxes and I figured the dust cover was in another box.  I only discovered later that day that underneath where the turntable base was, there was also the dustcover, wrapped in bubble wrap, and it had been broken in three pieces presumably by the weight of the SL1000 base and motor.  Unfortunately, at that time, I was so horrified I didn't take pictures of it until much later.

Sunday, March 23, 2014

I have seen data sheets for Burr Brown PCM63 and PCM1702 that call them Sigma Delta.  Never the PCM 1704.  Maybe marketing changed, but all three of these dacs are are now known as PCM dacs, not sigma delta dacs.  They all feature dual collinear dacs summed together.  These collinear dacs uses the same ladder in opposite directions, hence canceling out error.

Here is the official data sheet on PCM 63.

I actually have a DAC that uses these, an Aragon D2A2, currently not working (some electrolytic in power supply failed and leaked).  I loved the fact that it has HDCD, but I hated the fact that the input selector goes back to position 1, which I couldn't use, after every power outage.  I had it wired in for the disk player (then, a Denon 2900 which lacked HDCD) and the Sonus input (to de-code HDCD discs), and the Dish receiver (to get Dish music).  The Dish connection (via Toslink) often sounded underwatery for some reason, so I quit using it and had to run Dish audio through an isolation transformer to fix a ground loop I had then.  (Little did I know, my whole house was ungrounded then.)  Getting HDCD was always problematic too (of course Sonos had to be on max level).  I rarely did the Sonos thing with it, too much hassle.  And before I knew it (a few years after I got it in 2007 or thereabouts) it died (20012?) with leaky capacitor in the power supply.   My Aragon 8008 BB from that same era is still doing fine.

Anyway, I do need to get D2A2 fixed.  It really is nice.  And now I know ladder dacs are interesting.

I saw StephenSank on bragging that his Pioneer DV-AX10 uses PCM1704, but that's not what the Pioneer website says, so I posted that to this blog about 1704's.  It turns out earlier ones, or something like that, used 1704's (it's shown in a 2000 service manual) but later Pioneer switched to Analog Devices (as one person saw when opening one).

Here's a great list of which DACs us which DAC chips.

Most of those are nearly unobtanium, or ultra expensive even now.  I saw the Levinson 360 going for $3100, and the Wadia's sell for still more.  Now, 14 years later.

An assemblage 3.1 sold for $1350 late last year.  Those are nearly unobtanium, it looks like.

On Sunday I bought an Onkyo RDV-1, which features two 1704 dacs, and apparently clock developed by Apogee.  It has SPDIF coax and toslink inputs, and can be used as a DAC.  One reviewer in 2012 found it better sounding as a DAC than his Schiit BiForst.  I am hopeful that it will at least allow 96kHz input (most likely it will not allow 192kHz input given the date of manufacture, but 96kHz spdif was common by 2000, and especially since Onkyo went so far as to get collaboration with Apogee, there is a good chance the input will allow 96kHz.).  Details have been hard to find on the web.  This was a DVD-Audio player made to the highest standards to compete with the top multichannel players of Sony, Denon, Toshiba, and others.  It was replaced far too soon with the RDV-1.1, which added SACD (what I now call modulated noise) capability, and the collectible 1704's were replaced with early sigma delta's by Wolfson.  The 1.1 lacked the external inputs too.  Onkyo apparently bowed to market pressure and the Sony SACD juggernaut.  Here's a blurb from the blog linked above:

Integra Research's RDV-1 DVD, DVD Audio, and CD Player
05/14/01 - The Integra Research RDV-1 is a THX Ultra certified DVD player that combines professional audio and video circuitry features to extract the ultimate performance from DVD, DVD Audio, and CDs -- including CD-R recordable disks. It can also function as an outboard D/A converter for other source units. The RDV-1 was designed from the ground up to set new standards for DVD performance and quality; the D/A converters and power supply alone make this product stand out from the competition. The RDV-1 uses 192kHz/24-Bit DAC to provide the most accurate DVD-Audio playback possible. The DAC uses a Vector Linear Conversion (VLC) system with a low jitter Master Clock developed by the professional audio firm, Apogee Electronics, of Santa Monica, CA. The Apogee clock all-but eliminates jitter and provides for the highest quality digital conversion available. The Vector Linear Conversion system completely eliminates the ""sonic unevenness"" inherent with conventional conversion methods.This low-jitter digital clock circuit was first developed by Apogee for the professional music recordingindustry, and is at the very heart of the best equipment used to make the master recordings for music. Now, Integra Research and Apogee have used the same technology to play these recordings back at home. Jitter is the measure of the lack of rhythm between digital sound samples. Unfortunately, the human ear and brain are very sensitive to these tiny timing irregularities. Jitter of just a few nanoseconds can compromise digital audio performance by interfering with the brain’s ability to perceive a stereo soundstage. By using the Apogee clock, the RDV-1 minimizes jitter and insures each digital sample arrives in perfect step with all the other samples. With all the digital signals zipping around inside a DVD player, there is a lot of potential for these signals to go where they do not belong. To circumvent these problems, Integra Researchhas developed high-quality dual power supplies to provide inherent DC stability and ensure that no traces of digital artifacts enter the audio paths or analog ground. The Integra Research RDV-1 is also state-of-the-art when it comes to video. It has progressive scan video output for a smooth, flicker-free image, and compatibility with digital-ready TVs that can upconvert video signals. The video playback system uses a 27MHz/10-Bit video D/A conversion with four times the accuracy of conventional 13.5MHz/8-Bit systems. In addition to a full complement of optical and coaxial digital outputs, the RDV-1 has a multichannel analog output (DB-25) for simple single-cable hookup of multichannel applications.

96kHz is all I need right now, since everything else I have runs at 96kHz, except I can use my Lavry AD10 to resample higher rates from my Denon 5900 or Oppo back down to 96kHz, so it can run through my Tact and 2496 DCX.

I figured out finally how I can get digital output for the midrange to run through a real ladder DAC (and not the reasonably good, but still sigma delta AKM, inside the DCX.  The answer is, bypass the DCX for the midrange!  I wonder why I had never thought of this before.  I can use a DEQ to create whatever crossover I want by combining parametric EQ's !  And the 2496 DEQ does have digital output.  And it can crucially add the needed time delay so I can time align my speakers, as before.  Actually, one could create a super DCX this way with three DEQ's.  But I think it's only worth bothering with for the midrange (which is 80Hz to 20k on my Acoustats, so it's really almost everything except deepest bass and super extreme highs), because it is a bit more hassle than using the DCX, as well as being more expensive (drop in the bucket compared with other solutions, like the $1000 digital output upgrade for the Behringer, which is no longer listed on a mod website, and I wonder if it handles the muting that the DCX needs properly).

So with my spare DEQ, and the new Onkyo, I can upgrade to full ladder DAC in the midrange.

Friday, March 21, 2014

PCM converters

Dan Lavry sets the story straight regarding PCM AD converters.  First of all, non-sigma-delta converters are PCM (or that's the term he uses).  Sigma delta converters can be 1-bit or multibit.  All converters were at first PCM, then 1-bit sigma delta appeared.  1-bit sigma delta had problems, so it has been almost entirely replaced by multibit sigma delta converters.

The Pacific Microsonics Model One and Model Two were PCM converters.  NOT sigma delta.  The actual AD unit in these (?both?) were Analogic PCM units tweaked by them.  The DA they used was from Analog Solutions and designed by Dan Lavry.

Lavry's DA924 converter is PCM, as is the Lavry Gold.  They are not a straight forward architecture though.  They were an improvement on earlier designs he did for Pacific Microsonics, Levinson, Wadia, and others.

Lavry now (2007) thinks multibit sigma delta are OK.  PCM still has advantages, but it is very expensive to do correctly, with hand calibration and the like.

The output of a 1-bit modulator can be copied straight to DSD, or it can be decimated to PCM.  But 1-bit modulators were replaced by multi-bit modulators.  Multibit sigma delta modulators require either an downsampler (AD) or upsampler (DA) to complete the system.

The Analogic AD used by PM was similar to Lavry's ZAD-16 made by Analog Solutions.

Whenever one tries to criticize converters based on specific things, such as architecture, he gets very testy.  Reminds me of the late James Bongiorno, who got testy whenever I tried to ask him any questions.  But we cherish our great designers regardless of personality, and Lavry is the best.  Meanwhile it seems to me completely fair to say 1-bit anything is total crap, multibit sigma delta is better, and PCM done right is the best.

Sadly, the unique (PCM with auto calibration) DA924 converter is discontinued.  So perhaps that is probably why Dan Lavry got testy and refused to say unambiguously what I just said.  He may have already decided to move on to multibit sigma delta, believing it to be better.

But designers are not always the best judge of their own work.  WRT James Bongiorno, while he may have been a brilliant designer, emulated by others, I think many of his designs weren't that good in totality.  The Sumo Nine was one of his favorites, but it had a fatal flaw IMO, it used fans (as did the original Ampzilla).  Likewise nearly everyone who isn't a Bongiorno fanboy thinks The Charlie wasn't the best tuner at the time (I don't have one of his personally adjusted ones with rack handles, but I've heard lousy reports regarding those also).  With Dick Sequerra, his best tuner was likely not the one with his name, but the Marantz 20b, and FM has continued on far longer than he ever anticipated.  And so on.  How can I be the idiot that I am, and know these things that the genius designers didn't know?  I don't know.  All I do is listen and think.

I think designers of analog converters have been far too preoccupied with voltage accuracy.  The most important thing, most likely, is in preserving timing.  And as James Bongiorno told me point blank, the truth is not in the measurements.  It is in the math.  I now believe he was correct.  I was asking him what measurement I could do to find the weakness in a pulse count detector.

I think if I had the money, I would try to get the DA924.

Wednesday, March 19, 2014

Short Comment on DSD, and supersonic frequency response

My comment to another comment following Mark Waldrep's interview of John Siau:

The 6dB SNR would apply if there were no filtering, it is the wideband delta sigma noise up to the nyquist frequency (1.4mHz for DSD x64). While it might seem at first glance that DSD has wider bandwidth than any PCM, because of the astronomical sampling rate, that turns out not to be true, because of the huge need for filtering the high frequency noise, which doubles in each higher octave for delta sigma modulation. I applaud you for doing critical blind A/B testing and honestly reporting what you have not heard. More people should do that before claiming huge benefits to technologies like DSD, and therefore saddling us with cumbersome and restrictive formats that do not allow user digital signal processing, something I do in all my systems, and the digital signal processing has immediately obvious audible benefits (I do dsp in 24/96), unlike these differences between digital formats, which I feel are overhyped.
As for myself, I have not done so much blind testing, but it is my belief, from JAES published research, and casual listening, that supersonic frequencies ARE at least somewhat important. My best systems use super tweeters with response to 40kHz. So I do care about extended frequency response and low noise to at least that point. Furthermore high frequency loss and noise in an audio system is cumulative, so it has been reasonably argued that any given component should have response well above 20kHz, and low noise out there as well. I think it is wonderful to have audio files in 24/96 and 24/192 for this reason, among others.

Monday, March 17, 2014

Goosebumps again from Vinyl playback

It's been a long road (more than a month now anyway).  I finally got around to mounting my new-in-box Dynavector Karat 17D3 cartridge to the Mitsubishi LT-30 turntable.  This was actually less tricky than mounting the Shure M97xe cartridge last weekend in many ways.  But as I was more careful, and also needed to attach new headshell leads, it took longer.  I used the Sumiko headshell I had used on the Sony PS-X800 back in the 80's and early 90's (after repair, the Sony wouldn't auto balance with such a heavy headshell).  The headshell leads were mostly too tight and hard to get on (especially onto the cartridge).  Some were too loose to fit on the headshell, and I carefully pinched them down to fit snugly so as not to be easily moved.

One of the harder parts was setting the overhang.  I first did this by comparison to my 17D2 cartridge mounted on a Stanton headshell.  Then I put a tiny dot of removable adhesive on the turntable beneath the stylus of the existing Shure cart, and then verified it with the Grace that had come mounted with the Mitsubishi.  Then I verified using blank anti-skating test record.  The first time, the cartridge seemed a bit jittery.  I played a record and thought it didn't sound right.  So I got out the Shure protractor, and decided I had mounted the cartridge too far back.  I next moved the cartridge to the edge of the headshell.  Then about 2mm beyond the edge.  Then back a few times.  Finally, I ended up with the cartridge just about 0.3mm beyond the edge of the headshell.  This is at the limit of my ability to set the protractor straight on the turntable.  On the blank record, the stylus does not move when set down and has no jittery movements.  It still could be off by a tiny amount, but it seems good for now (until I have a different kind of test).

I went straight to the most troublesome LP, the one that at first made the M97xe sound horrible (until I disabled the brush and set the volume levels in pre amplification correctly to avoid overload, but even then it sounded more edgy on the M97xe than on a Shure M55e, which I think is due to excess capacitance in the tonearm cables--they measure about 430 pf whereas the M97xe requires about 275 pf).  This is E Power Biggs plays Bach at the Thomaskirke.  This record was heavily worn back from my college days, using a Shure V15 type III and later ADC XLM Mk2 improved, on a Dual 1209.T

(Update: the cheap TC-750 moving magnet preamp I bought for testing purposes, actually quite good I think for the $55 price, has a pretty high amount of input capacitance, 220pf.  Actually, that would be about right if the turntable had low cap cables, then I wouldn't have to solder in a capacitor, otherwise I would, so it's actually a good design choice.  But combined with the 430pF capacitance cables, the cartridge is loaded by a huge 650pF capacitance.  That's probably creating a 3dB bulge right in the sensitive 2-10 kHz region, centered right at the 6kHz screech zone.  So it would be expected to sound a bit harsh.  Meanwhile, that has less ill effect on an M55e, though it should be noted that cart is on a different table, the Miracord, whose cables I haven't measured.  Other specs for the TC-750 are 20-20kHz +/- 0.5dB RIAA, THD less than 0.05%, S/N 85dB, 3.0mV input sensitivity with 20dB overload (i.e. max input 30mV, which is ok, though I generally like more headroom), max output 1.8V.  Pretty hot for $55, I remember when you could expect +/- 3dB and 3% THD for that price.)

The incredible transparency of the Dynavector 17D3 gave me goosebumps.  I was hearing the flaps of the little pipes open, and even the echoes of the flaps opening.  The spaciousness of the recording was revealed for the first time (I didn't hear it with either Shure).  It was like I was really being there.  There was still quite a bit of groove distortion, probably from wear and dirt, but no more than the M55e, and with far greater transparency.

I got goosebumps from other records also.  I believe the 17D3 has eliminated almost all of the fluttery sound I was getting with the 17D2, the fluttery sound that I now know was coming from the tonearm cartridge resonance (because it went away with the Shure M97xe).  I hear artistic pitch bending, such as in Rick Wakeman's Journey to the center of the earth, and major and minor keys make sense (as they didn't with the 17D2).  I still fear there is a little less than rock solid pitch stability, as I seem to hear with the Lenco and Shure M91e.  I'd have to A/B with the Shure M97xe the LT-30 to be sure they aren't about the same or not--the Shure might have been a tad better for some reason (more damping?).  Might be good for the Mitsubishi to have a complete cap replacement and recalibration.  It might be that the Lenco will always have better sound.

But as of right now, I have respectable vinyl playback, finally, for the first time in many years.  The Dynavector is clearly more transparent than any Shure cartridge I have, and in the Mitsubishi arm it tracks perfectly with well adjusted automatic operation that I don't have to worry about.  (I worry about arm tilt on the Lenco because of the semi-broken condition of its tonearm, which tilts easily and has no working anti-skate and the stylus pressure adjustment is unstable thanks to drooping counterweight.)  I don't need to fear damaging my records with a brand new Dynavector tracking well in a linear tracking arm.

Sunday, March 16, 2014

Hirez PCM is better than DSD

PCM 24/96 and 24/192 are better

So says John Siau, Director of Engineering at Benchmarks Media, one of the makers of the one of the best DSD and PCM dacs, interviewed by Mark Waldrep of AIX recordings, some of the best recordings available on DVD-Audio (I got my first recording from them something like 10 years ago) and now Blu-Ray and hirez downloads as well.  I'm making this my top link because John Siau gives some of the best technical explanations I've seen anywhere, and I believe he is 100% correct in what he says here.   Some of what other people I link to below I may agree with but leave me feeling a bit less than 100% certain.

John says Benchmark supports direct DSD processing basically because some people have those files, and if you do have a DSD file the direct processing of DSD is slightly (though just very slightly) better than conversion of DSD to PCM first.  He stresses however that the DSD->PCM conversion is very benign, nonetheless.

But John says he does not endorse DSD, and believes that people should migrate to 24/96 and 24/192 because they are better in every way.  He also points out while DSD makes an ok delivery format for high rez digital, somewhat better than 16 bit digital anyway, and it's all most people with most systems need, it makes a lousy professional recording format.  And the main reason it makes a lousy recording format is that any kind of editing or mixing or even level adjustment requires conversion of DSD to PCM first, at least in some form, and then back to DSD if that is how you want to distribute the recording.  And then the big loss is in the conversion of PCM back to DSD, that conversion is not so benign, he says.  He gives way more technical detail than I do here, and also debunks the legendary "noise graph" Sony used to sell DSD, wherein the noise level of PCM formats is shown as a straight line higher than the noise level of DSD, whereas in reality the noise spectrum of 24/96 and 24/192 is below that of DSD (64x) everywhere, not just at supersonic frequencies (where the noise of DSD takes off like a rocket).

At least one DSD'er appears in the comments, arguing by authority of someone he knows who says DSD is better.

DSD = Never Twice the Same Digital Recording

Here's the best link I've found so far with a simple but visual comparison of PCM and DSD and the allegedly better impulse response of SACD which is actually worse.  Basically, a delta sigma audio system like DSD generates a huge amount of noise at higher frequencies.  The "Noise Shifting" that DSD uses to hide this fantastic amount of noise is a mathematical feedback loop which cancels out high frequencies generally (it can't actually discriminate between high frequency signal and noise).  In the process of canceling out the high frequencies up to the Nyquist limit of 1/2 the sampling rate (1.4mHz for 2.8mHz DSD) this feedback loop moves the 1-bits around, and this creates noise in the time domain, also known as jitter.  (I explore this further below in the section with Jitter in the heading.)  This jitter can hide the fact that if you overlay enough DSD recordings of the same thing, such as a 10 kHz square wave, what you ultimately see looks very close to 88.2kHz 20 bit PCM.  It's just that every time you make a DSD recording from a square wave, you get a different portion of the infamous (but actually harmless) ringing at the top of a square wave in PCM.  So perhaps DSD should be called "Never Twice the Same Digital Recording" because the exact recording you get varies from one DSD recording to the next.  (The playback, however, will be mathematically identical each time.  So the randomness is no fun for the listener, who is actually being cheated out of getting the whole enchilada, no matter how many times he listens to it.)  DSD also gives you nothing like the high frequency response you might expect from the sampling rate.  While you might expect frequency response to 1.4 mHz from a 2.8 mHz sampling rate (not exactly DC to light, but getting up there), what you get is actually limited to about 40kHz, not any better than 88.2kHz PCM let alone 96kHz or 192kHz.  Meanwhile, high resolution PCM formats potentially give you frequency response and low noise all the way to just below half the sampling rate. It's been proven that we are affected by such high frequencies, and/or the slew rates they embody, even when we can't hear them as tones.  (Since this link above is available only through the Wayback website, and no recent comments, I wonder about it being 100% true, but you can see the corroboration below.)

Another Format War Now, just when PCM has gotten so good?

Nowadays we have wonderful PCM systems available, 24 bit 192 kHz recorded music is available just about everywhere, on DVD-Audio discs, Blu-Ray discs, and high rez downloads.  And incredible 24/96 or 24/192 playback equipment is widely available at low cost, $100 and up.  And PCM is suitable for endless digital signal processing (DSP) options, from digital preamps, to parametric and graphic equalizers, room correction systems, digital crossovers, vintage simulators, etc.  Working with PCM inputs and outputs, these processors can be noiseless and distortion free.  Note that it's basically impossible to do digital signal processing directly on DSD; it has to be converted to PCM or analog to do that.  My own systems rely on digital equalizers and crossovers working with PCM at 24/96.  It would be impossible to assemble the hybrid speaker systems I have without these digital processors.  I also love the high resolution PCM recordings that I have obtained on DVD-Audio discs.  As recently as last year I obtained the DVD-Audio anniversary edition of Lark's Tongue in Aspic from King Crimson, featuring 24/96 and 24/192 PCM masterings, and it has become one of my favorite recordings.  This was released in 2012, long after many in the audiophile media declared DVD-Audio dead.  In fact, many declared DVD-Audio dead before it was even introduced in 2002, and I can't count the number of wonderful DVD-Audio releases I have picked up since then.

But just now that we have so many wonderful PCM options, Sony and a particular cadre of super high end equipment and software purveyors and studios are back to pushing the old 2.8 Mhz 1 bit DSD from 1996 as being somehow better than all PCM, while many of the most highly respected audio engineers and enthusiasts (including me) long ago thought 2.8 Mhz DSD was just barely better, if at all, than Redbook 16 bit 44.1 kHz digital, and very inefficient as well, as well as not being amenable to digital signal processing.  If you ask me, this is a disaster for good recording, and adept home audio enthusiasts like me because another format war killing off high resolution PCM (now available on Blu Ray as well as DVD-Audio discs, as well as downloads), and wasting the time and effort of equipment manufacturers, is the last thing we need.

My Experience, and that of my friends

Before buying my first Sony SACD player, but just based on technical articles I had read (see below), I believed SACD actually would be inferior to Redbook digital, and was just a marketing scam by Sony to keep their ability to extract license fees intact after the expiration of CD patents, and keep music locked up in unrippable discs now that the public had realized CD's could easily be copied and uploaded into digital servers.  I bought my first DVD-Audio player as soon as I could, a Toshiba 5700, having great hopes for high resolution digital on DVD-Audio.  Only later did I grudgingly buy a Sony SACD player so that I could take advantage of the greater variety of claimed-to-be-high-resolution discs in their native format.  Back in those days, I fed the analog output of my Sony SACD player straight into an all-analog preamplifier with a system having no digital processing.  I continued to seek out the DVD-Audio versions of everything I could first, but had to accept that far more discs were available in SACD, as a result mainly of the possibility of Hybrid CD/SACD discs.  I did feel after all that SACD was somewhat better than Redbook digital, just no where near 24/96 PCM.  In all this time, only one recording on SACD has seemed to reach about the same heights as 24/96 PCM on DVD-Audio, and that is the SACD version of Santana's Supernatural.  But I have found countless DVD-Audio discs (and such variants as the DVD-Video DAD's from Classic Records) that I though were stellar, far better than anything I could imagine hearing on CD.  Yes I have been very unscientific because I got tired of doing endless listening tests long ago, and came to never trust the outcome of a small number of tests anyway.  So why should you believe my listening tests?  Of course you shouldn't, but many people seem to be interested in my observations and thinking anyway.

Meanwhile, many serious audiophile friends of mine (including one audio manufacturer, and another veteran audio engineer) have never bought into the notion that DSD/SACD was better than Redbook, indeed considering DSD/SACD to be inferior to Redbook CD digital.  I must add that these friends ubiquitously used the analog outputs of the latest generations of SACD players available at the time, almost always made by Sony, into their all-analog preamplifiers.  I must add this qualification because many DSD proponents are now claiming that "people who criticize DSD have never heard it properly" because in their imagination the detractors were so dumb as to take the PCM outputs of SACD players into their DAC's (or worse, use the Home Theatre setups on DSD, that requiring internal conversion to PCM) and that of course would not preserve the sanctity of DSD, as it would have to be translated to PCM before decoding.*  But none of my friends have been that stupid, and none have bothered testing SACD players in home theater setups either (having DVD players dedicated to that task, if they even bother with Home Theater).  They brought home the latest Sony SACD/CD players, plugged the analog outputs into their analog preamplifiers, and tried them out, using Redbook and SACD's, most often on the same exact SACD/CD players, though sometimes with different players and dacs attached to those other players, and concluded that the CD layer sounded better than the SACD layer (and being careful to find CD/SACD that were mastered identically, which was most often not true), or that the CD layer on the hybrid disc seemed dumbed down compared to regular CD's which could be purchased at the same time, and that those plain old CD's sounded better than anything on the SACD or hybrid SACD.  And then, back went the expensive latest generation SACD player to the store or friend it came from.  What actually bothers me is that some of these friends have never even tried hirez PCM on DVD-Audio.  They just tried SACD because the herd was going there and they wanted to try it themselves.  And there were so few DVD-Audio discs available, and almost never from local stores.

That has not been my experience, which I described above, though to be honest I haven't done much side-by-side testing of CD's and SACD's.  I just play SACD's, on my Denon 5900 or Oppo BDP95, pretty much assuming SACD or DVD-Audio would be at least slightly better, and that has been my overall impression, from listening to all the discs in my collection, in other words casual testing if sometimes done through serious listening.  And I've always used the analog outputs, and never external DAC's, which I feel are not worth the upgrade from the fine disc players I have, which I believe have better circuitry than most DAC's ever made, and don't suffer from jitter that may arise through external digital connections, especially ones that are not AES/EBU or better.

(*Though I suppose you could say that some particular DSD DAC available today is better than all the ones these friends of mine have heard, so much so that the (alleged) superiority of DSD from an SACD over an identical CD was not then apparent.  Yes you can always make such arguments, ad nauseum.  Frankly I don't think huge differences even exist among PCM DAC's since about 2000, or maybe even 1986, even from the DACs simply built into the better quality CD/SACD players which have typically used the best available DAC chips if not the most overbuilt associated circuitry, the technology and measured performance are only marginally different, and among DSD DACs since the SCD-1, which was a pretty much all-out effort by the inventors of SACD.  I have no time or interest in testing each latest thing that has come out.  I prefer more obviously fruitful and less expensive endeavors, such as fine tuning my digital crossovers, or actually listening to music.  Or pointless but personally satisfying endeavors like posting to my blog site.  However I'm fine with people playing their latest and greatest toys in their homes and audio society meetings and conventions for my amusement.  Anyway, before I bother taking your argument about so-and-so being the best whatever, so much better it invalidates all previous tests on the same kind of thing, you'd also have to provide some sort of reason that makes sense to me, or if you have the unit in question right and hand, you can play it, and I'll give a suitably ambiguous observation, if you want one, or maybe even if you don't.)

The Technical Literature

Here's the first paper by Lipshitz and Vanderkooy blasting 1-bit Sigma Delta conversion.  Lipschitz is one of the most well regarded audio engineers of all time.  Another very knowledgable audio engineer, David Rich,  summarized those ideas and others in a harshly critical review of DSD/SACD in 2001 that was (surprisingly) published in Stereophile.  John Atkinson posted a one sentence follow up to Rich's ironclad denunciation (showing among other things that 1-bit DSD would generate idle tones and other bothersome forms of distortion) with the claim that DSD was not, in fact, an entirely one bit system, as he had apparently heard from Sony after informing them of the critique.  However, I now wonder if that disclaimer by Atkinson was at least partly incorrect.  Perhaps some DSD recorders use a multibit system, but it seems like that hasn't necessarily always been true, and virtually all DSD proponents are still talking today about DSD as if it is actually a 1-bit system.  If DSD is a 1-bit system, then all the horrible things Rich, Lipschitz, and Vanderkooy talked about are indeed wrong with it.  If DSD isn't a 1-bit system,then it requires decimation and interpolation filters just as PCM does, and there really isn't any magic "less processing" required for DSD, only a very inefficient means of encoding audio information so that more bits actually yield less information than Redbook PCM (see Arthur Salvatore's opinion discussed below--and note that while Salvatore clearly states he is not a digital expert, he is actually echoing some of the arguments made by the the serious engineer David Rich).  DSD promoters seem to want to have it both ways.

After the paper by Lipschitz and Vanderkooy, some engineers employed by Philips responded in JAES, and Lipschitz and Vanderkoy respond again, and so on.  This continued until about 2003, by which time Sony wasn't pushing SACD much anymore and had moved on to preparing the battle royale over high definition video formats, which they ultimately won with a frequently alleged to be inferior system, but by getting a majority of movie studios on their side.  You can read about the intellectual battle over DSD in the Audio Engineering Society in the Wikipedia entry for Direct Stream Digital in the section entitled DSD vs. PCM.

While I'd recently become hopeful that DSD was a solution to the limited time resolution of PCM (see earlier posts), I now strongly believe that DSD is not the right solution.  Not only does DSD generate large amounts of noise (which is shifted to higher frequencies…it does not "go away") in the amplitude domain…it generates noise in the time domain because of the noise-shifting.  It doesn't actually have frequency response better than PCM, in fact it's substantially inferior to 24/96 in bandwidth.

A Blind Test

Listening tests at the River City Audio Society meeting for March 2014 were inconclusive for the group (compared with 16/44.1 digital) and very much for me as well.  I had not a clue which was which.  In contrast, a year before I clearly heard the difference between 16/44.1 and higher resolution digital formats like 24/96.  In that test,  I got every identification that I thought I got correct correct (all but one of the like-vs-like comparisons, and I don't count the mastering beauty contest last test).  Unfortunately, the DSD vs PCM tests were likely all beauty contests anyway, with the sonic differences coming entirely from mastering or level differences.  But even then, I would have expected the superior sonic resolution, if there were one, of DSD to shine every once and awhile.  It didn't for me.

PCM continues to improve

It may be that whatever benefit DSD has could be duplicated better simply by having higher sampling rates, which are already available, such as  32bit/384khz PCM.  PCM has had a history of continuous evolutionary progress.  In DSD, one of the most highly regarded units is the very first one, the SCD-1 from Sony.  Now there are 2x and even 4x DSD recorders available.  But that's just catchup to the progress PCM has made since 2001, and at very high price levels

PCM also has many benefits, it makes possible wide ranging DSP such as speaker and room correction (which I use…manually tuned not automated), digital level control, and the like.  For those recording music (which sometimes includes me) it is far easier to work with.  To my knowledge there is only one (stratospherically expensive) DSD editing system.  Virtually all DSD recordings are edited or processed either through PCM or analog.  I think most of the recordings we listened to at RCAS were originally recorded in analog.  There are now systems that can losslessly edit (without losing the DSD magic) DSD files by up sampling to an even higher resolution format known as DXD, which is essentially 8 bit PCM at 2.8Mhz.  Such equipment is now stratospherically expensive.  One wonders if the game being played now isn't intended to rub out all the small producers, leaving only Sony and a few majors who can afford the mega priced production equipment.

OTOH, perhaps the "randomness" of DSD, noise in both the analog and time domains, is a good thing.  Life is random too.  But I can't argue that seriously here (but I will take a stab at it later…I have a concept which I feel is far superior to DSD, using only PCM systems and analog.)

Jitter, you ask?  DSD is made of Jitter

Well as I stated above, the "noise" in the time domain that "noise-shifting" creates (recall that noise shifting works by moving the 1-bits around to start filtering at about 10kHz) is also known as jitter, but somehow the J word is never mentioned among DSD enthusiasts.  They don't have the numbers.

I have often noted that some of the best research published on jitter shows that it doesn't become audible until about 10nS or so, which is like 50 times the jitter levels in the better digital equipment, which is more like 200 pS or less.  (I don't necessarily consider this the final word on the audibility of jitter, but it did represent the state of the art audio science in the mid 1990's.)  The very best equipment can get down into double digits (actual clocks can get down to single digits or less, but we don't listen just to the clocks).  So I think jitter may be overrated in importance, used more as a pseudo-explanation to cover up the fact that people don't really do controlled blind tests, but need to explain their gut feelings about why one CD player is better than another, and since the normal specs don't seem to say anything meaningful, audiophiles have latched onto jitter as the ultimate cause of all the bad things they think they hear, in unscientific listening tests, when in fact they don't really hear differences between digital playback gear, they just generally hear the lack of ultimate resolution in all digital sources, I think, maybe, and I am agreeing which Arthur Salvatore on this resolution thing (linked below) though this is completely unproven, it has always made intuitive sense to me that digital audio systems reduce noise at the expense of actual resolution.  What I mean is that the difference between 270pF jitter (a dCS stack from 10 years ago measured about this well) and 225pF jitter (that's what a Sonos system measured--and I found it mind blowing that an inexpensive network player measured better than the dCS stack, when lowering jitter was one of the reasons for a dCS stack) probably doesn't mean much.  But when you start getting into the nS of jitter, you definitely have cause to worry.

But what about DSD?  Well I started looking back at Stereophile tests, ,and starting from the SCD-1 and other CD/SACD players, jitter is not measured for the SACD layer, only for the CD layer.  John Atkinson gave the excuse that his Miller Research Jitter Test only comes on a very carefully made low jitter CD-R.  So I just did a search for SACD and Jitter, and guess what, he did graph a sort-of jitter test for the Sony SCD-XA777ES player, one of the best SACD players ever made.  This player had a very respectable 171pF jitter using the CD-R.  Using the Sony 11.025kHz SACD tone test (this is a rather poor test of jitter, actually, because it's not an inter modulating signal as on the CD-R) it shows a "noise" floor much higher for the simple tone SACD playback than for the CD-R intermodulation.  JA says the SACD playback noise floor was elevated 10dB.  Now how can SACD have a higher noise floor with it's alleged 20 bit resolution compared with a mere CD?*  Hehe, I can't help but believe what it's showing is the jitter, but because of the way SACD works it doesn't produce jitter distortion in spectral spikes, but rather an elevated broad spectrum across the entire audio band.  JA doesn't give us any estimated jitter number for it, but obviously it's significantly higher than for the CD playback.  Now 10kHz is just where the noise shifting is starting to kick in.  Imagine how bad it would be at 20kHz, or 30kHz--which SACD players are generally considered as having response to.  And the fact that this "noise" covers the entire audio band can't be good either.

(Note that at 10kHz, the background noise level for CD and SACD is identical.  But in the presence of the 11kHz signal, the SACD noise floor rises 10dB higher.  That's the awful sound of dynamic noise, noise that rises and falls with the signal.  Such things sound horrible, like old fashioned noise gating systems.  Even though SACD does not actually have noise gates, it seems like many noise reduction schemes, including the old Pulse Count Detectors from Kenwood, have a tendency to have dynamic noise which sounds as if there were noise gates.)

Whatever it's sampling rate, or high frequency response, I believe SACD simply doesn't have time resolution at high frequencies.  It smears them in order to have respectable measured noise levels.  And the high frequencies are where we need that time resolution, because the high frequencies are where the detail exists (the tiny curves on an oscilloscope are the high frequencies), and I believe we are affected by that resolution or lack of it at frequencies higher than we can hear as tones.

(*Actually, though, at 10kHz, the "noise floor" of the XA777ES in CD mode is the same as it is in SACD mode, as is typical for CD vs SACD.  That's where the two noise curves cross over, and the noise of SACD just takes off, despite the noise shifting.  Still, the identical background noise level at 10kHz doesn't explain why the SACD measured 10dB higher spectral noise while playing an actual 10kHz signal.  What would this mean in terms of Jitter?  I don't know how it scales, but I'd think at least 3.3 times and maybe more than 10 times more.  So at a minimum the SACD playback had 500pS jitter, or 2nS, or maybe even higher as the noise is spread across the entire spectrum.  And if you really tried to force higher jitter with a combination of tones, for example, you might get well into 10 nS and above  Interestingly, an Accuphase DP-85 measured over 4nS jitter through external input.  Was it "upscaling" to delta sigma modulation?)

We really don't have enough data here, and my knowledge is lacking too, but even the lack of data suggests something very wrong.  If SACD is better than CD, shouldn't it's jitter, a drum that audiophiles including Atkinson have been pounding for years, be less instead of more?

M.O.J Hawksford on the Meridian site says that "bitsream" (by which he means DSD and the like) is inherently more susceptible to jitter cause by intermodulation with the elevated high frequency noise floor.  But they don't give any number either, or mention DSD by name.

Opinions, opinions

Here's one of the seminal pro-DSD papers from 2001, essentially a response to the Lipschitz and Vanderkooy paper I linked above.

Yet another blog regarding DSD vs PCM at WBF.  This makes me wonder, many angels can dance on a single bit?  If DSD is as good as it's fans say, where are the blind tests proving it?  I'd be the first to say blind tests cannot tease out all that a person can hear, there have to be fairly big differences to hear differences in blind tests reliably enough.  But that is precisely what the DSD fans are claiming, big differences reliably audible in a studio environment.  Every conversion being audible to them.  Somehow real audio scientists don't get involved with those studios.  They probably wouldn't be liked much there.

Audiophile magazine reviewers are not going to say anything bad about DSD purveyors when they are large current and potential advertisers.  But many reviewers who write for their own pleasure feel no such restriction.   As well as manufacturers who don't use DSD anymore.

Linn Sondek is now calling DSD a good idea in 1999 but obsolete in 2013, followed by a 11 page discussion.  Not as good as 24bit/192kHz, which can be played on Linn DS players since 2007.  I have a number of Linn hybrid HDCD/SACD discs from the 2000's which sound reasonably good in either HDCD or SACD.  Either way, I must use them on a universal disk player since my servers don't support HDCD or DSD.  Haven't done enough testing to decide which layer I like better.  My general feeling has been that properly decoded HDCD is about equivalent, or better, than SACD, and in fact I have more favorite sounding recordings on HDCD than SACD.  HDCD played back without HDCD decoding is ok on Reference Recordings but mediocre on other labels, so if you haven't heard HDCD properly decoded, as it seems few audiophiles have, you haven't heard it, and this is incontrovertibly true.

Romy the Cat hates DSD with passion.

Arthur Salvatore (for whom I have very great respect, btw, more so than virtually any other audio reviewer, at fact value anyway--since I also respect John Atkinson but know he is very constrained by his commercial perch, so I read JA between the lines always and look at his measurements which I cherish, as well as Peter Aczel who has virtually opposite opinions on nearly everything--but see my comments on him below) hasn't much liked any form of digital, though he concedes it is superior to analog in many obvious ways, other than the lack of ultimate resolution which he believes is the most essential feature of good audio.  He believes digital will ultimately catch up, even if it hasn't done that so far, with higher sampling rates and bits of resolution to meet human sonic requirements--I completly agree with this!  He seems to greatly prefer CD to SACD/DSD, pointing out that in many ways CD's have more actual  resolution.  He points out that DSD encodes 64 different states at 44.1kHz, whereas Redbook CD encodes 65536 different states at 44.1kHz.  Then he also points out that modern deltat sigma DAC's for PCM also degrade the inherent information content and therefore resolution similarly, but no where near as much as DSD does.  BTW, he also recommends only disc players with or without external DACS.  His one test of an (unnamed, at the manufacturer's request) expensive digital server proved very disappointing.  I find that quite believable as I know digital servers use asynchronous and highly jittery connections like USB (gag!) and ethernet, as well as being internally jittery in many ways. I had not been thinking that way myself recently,  but reading him it makes sense.  BTW, the relative information content in DSD will actually increase at lower frequencies…so at what frequency are they the same?  43 Hz!  Below 43 Hz, fwiw, DSD will have more information than Redbook CD played by a non-oversampling DAC.  Unfortunately for DSD, most of the music is above, not below, 43 Hz.

A fixture on the PS Audio blog, Elk, finds more disadvantages than advantages to DSD.  He says that the purist form of DSD is incredibly rare.  That would be the recording that stays in DSD all the way.  DSD can't be edited or processed because there is no headroom (the full bandwidth S/N is 6dB).  So in the real world, conversions are done between DSD and PCM, and those conversions cost more than the alleged benefit of DSD.  Further he describes conversion to analog for editing purposes as very lossy.  He says the only huge advantage of DSD is the ease of implementing an acoustically transparent anti-piracy code.  He also explains how modern PCM and DSD are very similar, and that modern DSD ADC's and DAC's are really 4 bit (Romy says the opposite, that DSD has stuck with or gone back to 1-bit, when 4 bit DSD was better, but I believe Elk more).  He says that PCM has wider bandwidth and S/N, as well as being more easily edited (by the recording engineer or the consumer--I loved to read that, he's ok with us consumers having freedom to edit) and converted to other formats.  However he chickens out from saying hirez PCM sounds better.  He says decide for yourself.  Speaking of which...

Cranky veteran audio journalist Peter Aczel points out that neither high rez PCM nor DSD has proven to be audibly superior to correctly produced Redbook CD digital.  He says the clinching research was done by Meyer and Moran and published in the September 2007 issue of the Journal of the Audio Engineering Society.  He says that often SACD's or DVD-Audio discs sound better only because they are edited better.  He does in fact buy and review SACD's and DVD-Audio's for that reason.

I like Peter Aczel.  I like reading what he says.  I feel my disagreement is small, but he might not agree, and YMMV.  As far as I know, there has been no scientific proof that high rez PCM or DSD is better than Redbook PCM.  It sounds quite believable to me.  However, scientific proof is not all that we go by.  It depends on how you value type 1 and type 2 errors.  If you only want to use a higher rez format if it has been proven better, otherwise you don't want to waste your money, then you should listen carefully to what he says.  OTOH, if you think something just might be better, by a tiny amount, even though it might not have been, or might never be, proven to be better, and you want it, and you're willing to take the risk that it might not actually be better, and you have the money for it, then you shouldn't follow his advice.  And it's interesting that he doesn't always seem to follow his own advice.  He owns far more expensive amplifiers and disc players than have been proven to be necessary.  Sometimes he has rationalized this, along similar lines, that he wants equipment sufficiently better than the minimum requirement that he doesn't even have to think about it.  He tends to go for the stuff that's either measurably better or designed according to the most sound engineering principles, as he see's them (which is pretty much like most audio scientists and engineers).  That's not far off from what I do.

But I think it's very refreshing to realize, nonetheless, to realize that hardly any of the hoopla that audiophiles make big deals about has really been proven to be better.  The reason is that this gives me the liberty to do my own thing, and now worry about the people telling me I must do such and such, or I'm not with it, not a real audiophile, or whatever.

With regards to statistically valid blind testing, I think it's a very good idea.  It's hard work, though, which should not be underestimated.  It has to be done 100% correctly to be valid.  And that requires precise level matching, use of absolutely identically mastered material, and pre-determined number of trials to reach statistical significance (typically, this is about 25 trials, though if you were sure to get every trial correctly, 10 would do).  Interpretation of the results depends somewhat on expectation.  If you believe a huge, obvious, and incontrovertible difference exists, and you get p value well above 0.05, you should at least adjust your expectations.  OTOH, if you think the difference is actually quite hard to hear, and predict that you might not get it the first time,  a p value even as high as 0.2 could be "suggestive" and warrant further experiments (note that all experiments must be reported, not only the ones with good results).

I think differences that have been shown to be important in such testing should be top priorities.  Differences which lack such proof probably are smaller and less important, if they even exist at all.  We do know that correct frequency response is quite important.  We don't know that the omitting the interpolation filters, as DSD does (for the 1-bit DSD which may not actually exist now, if it ever did, as many I've quoted above say that actual DSD implementations are typically 4 bit nowadays, while the 1-bit mantras continue) is very important.  Being able to omit the interpolations filters probably isn't very audibly important, or Sony probably would have proven that it was, in statistically valid blind testing.  They had the means and motive long ago, and didn't.  So it's probably not a big deal, if it's even a deal at all.  And since the DSP devices I like to use require PCM, which with modern DAC's available to me require interpolation filters, I will continue not to omit interpolations filters, by not making my system into a pure DSD system.  And I will not feel guilty about it.  And I will not even feel I'm not a member of the What's Best crowd, simply on those grounds anyway.

Often audiophiles argue from authority on the one hand, (so-and-so says DSD is far better than PCM, and they use the whatsit system which everyone knows is the best) and/or then uselessly say you should listen for yourself (which is no substitute for scientific proof either, and the result of a single listening test is about as informative as a coin toss).  The knowledge that certain things have NOT been proven means that you or I are free to disagree (except over whether things have been "proven") and not feel cowed by self-proclaimed experts having far more expensive equipment, etc., then we will ever do.  The benefits of that far more expensive equipment has not been proven either!

16bit 44.1kHz digital better sounding than DSD ?

After writing much of above and thinking, my best answer is, I don't know, but I am beginning to think so.  For the last 8 years or so, I had been assuming that SACD was better than CD.  I didn't actually believe that until I got my first SACD player.  Then I let myself get taken in.  I haven't done serious testing, and I suspect it would have to be very serious testing indeed.  And very difficult to do, since the identical masterings are almost never available, even on the same disc.  And to be "identical masterings" they would both have to come from some technology superior to both (live recording, certainly, or DXD?) or at least identical (analog tape, which may or may not be better, though I actually believe it is in the highest resolution available, say 1/2 or larger tracks at 30ips).  If you simply take the DSD and convert to PCM, you are not hearing all that the PCM is capable of, because the initial translation or recording to DSD is one major limiting factor.  Making a DSD recording throws away at least half of the information that would ultimately go to the PCM, even in 16/44.1 !!!  Sony, which owned the masters for the pure and hybrid SACD's it made certainly had no incentive to do this.  One of my friends strongly believed the CD layer on hybrid SACD's was inferior to previously existing CD's, and I now think he was probably right.  Most likely Sony did in fact get the CD layer by conversion from the DSD, and that would not be "fair" to the CD layer.

Charlie Hanson has now made some tests available in which both the PCM and the DSD come from the identically same analog masters.  I think those are worth getting.

I believe it is actually true as Arthur Salvatore argues that 16/44.1 has more actual information.  DSD is a kind of lossy encoding.  But remember that half of the information in a 16/44.1 recording is above 10kHz, and the other half below.  DSD does an increasingly good job of encoding the information below 10kHz as you get lower in frequency, it merely does a lousy job of encoding the information above 10kHz.  I'm not exactly sure how to compute this, though my computation of the point where DSD and 16/44.1 have the same state information is 43 Hz.  That's the point at which they both directly encode 65536 states.  But there must be more to it than that.  Somehow the DSD is shown to have less background noise below 10kHz, and I don't quite how to quantify that, though it's clear the lower noise is a result of the "noise shifting", and it jitters the information below as well as above.  So if I said above that the noise shifting begins at 10kHz, that is incorrect.  Actually, the noise shifting is through and through.

Still, DSD has more time Resolution

Thinking about the high sample rate…it still occurs to me that 2.8Mhz DSD has more time resolution.  The very beginning of a sonic event is not lined up to the coarse grid of lower sampling rate PCM.  This is a breath of the original sonic even that gets through pure DSD, if such a thing exists.  OTOH, if you have a DSD converter based on 3 bits, say, then you have to divide the nominal sampling rate by that number to get the real time resolution.  Any conversion, even conversion to 386kHz sampling rate DXD, will squash that resolution back down to that point.  Then you've only again got a marginal increase in time resolution than DSD, and all the additional time distortion afterwards that DSD adds in the name of noise shaping.  So while the very start of an acoustic even can ideally have more resolution, it gets smeared after that.  Meanwhile, the regular time grid of PCM has a kind of coalescing and sharpening effect, as everything is combined to start at each sampling interval, whether it actually started a little before or less is lost.  However, there again, with sigma delta PCM dacs, and even through it's kind of fake, the sharpening is removed, and there may even be a bit of apparent decoalescing of multiple events all having to start at the same instant, but that decoalescing is fake, the information really isn't there.  Running PCM through a sigma delta DAC (preferably multibit, of course) will make it smoother, without any actual increase in resolution.

People who can watch NTSC on actual NTSC television, or multiscanning crt's like my Sony XBR-960--a true classic still worth having, will see it as far sharper than NTSC remapped to 720p or 1080i, despite those HD formats having more resolution.  And this added sharpness, mostly appreciated, is real  sharpness, not the fake sharpening by peaking the horizontal amplifier that sharpness controls generally do.  Real sharpness is always better.  Meanwhile the 720p or 1080i presentations of 480i will appear smoother, but it isn't the smoothness of something which isn't artificially peaked, it's the smoothness of blurred edges, and that's what the delta sigma conversion of PCM does.

[this post may continue to be edited and expanded]

Monday, March 10, 2014

Escaping flutter and edginess with Shure M55e

I just summarized my weekend discoveries at VinylEngine

Right now, a rig with M55e beat everything other rig I have. I think it's at least partly how good the m55e itself is. This is a used cartridge and it could very have original stylus from the early 70's, it looks genuine anyway.

This blew my mind. I was expecting to toss the used M55e on this Miracord 50H table into the trash, not my level of kit.

Other contenders included Dynavector 17D2 from 1985, my previous reference cartridge. Apparently due to damping system deterioration, with maybe some tonearm incompatible with the arm it's on now, it adds serious flutter, I thought the Mitsubishi LT-30 turntable it was on was broken. I see now that old cartridges may have issues other than stylus wear, though this cartridge has also been used off and on over 18 years, it never sounded bad on a Sony PSX-800 with biotracing to handle the resonance, when the PSX-800 worked. The 17D2 always sounds perfectly transparent and never harsh, but the flutter is intolerable.

An M97xe on the Mitsubishi eliminated the flutter, but was edgier than the M55e, without being any more transparent. On the right arm for it, I expect it would be better. But it brought back sad memories of the V15 III I had in the 1970's.

The M55e has a wonderful relaxed quality, and it just sails through everything, not being too picky about anything. I don't have the V15 III I had in the 70's, but I wouldn't be surprised that if I had both now, I would like the M55e better. I think this may be THE cart to have for old worn records, when you're not trying to get the nth degree of transparency, though it is pretty transparent, I'm sure a new moving coil would surpass it in every way, but I haven't got a new moving coil mounted now. When I had the V15 III back in the 70's I thought it sounded slightly edgy and closed in at the same time. Even back then, I wondered if the earlier Shure models which I occasionally heard weren't better, and if Shure was on the wrong path to transparency and I switched cartridges many times after that, never going back to Shure, and ultimately ending up with the Dynavector 17D2, which was stellar on the Sony PSX-800.

I'm wondering if this would actually accept a V15 type 1 stylus, since it was basically a V15 type 1 with lower QC. That might be the way to get higher quality stylus for it, but it might also lose some of the relaxed quality that makes M55e worth keeping.

Here's a thread by lovers of the V15 series cartridge at AudioKarma.  The V15 type one gets little respect, people say it started getting good with the Type III, exactly the one I love to hate.