DSD comparison to PCM.

You are correct - but I also found the rumor spread all over the net! Just found the truth in a old review - the 20400A used a BurrBrown 12 bit multiplying DAC (the DAC7541A) coupled to a discrete 8 bit DAC. Thanks for noting it! Surely it must be considered a true NOS! :)

Non-oversampling or what I call a ladder DAC, although this one was done in two segments. Still, no noise shaping, only deglitching!

The demise of Ultra Analog was one of the reasons that production of the PM Models 1 and 2 ceased. MSB are perhaps the only ones who still make a ladder DAC, although I've never had experience with theirs.
 
You are correct about the internals of the UltraAnalog 20400.

My source of knowledge about this part was Paul Frindle who was using them in his DAC design for what eventually because the Sony Oxford digital console. He had the stereo parts configured in mono as the standard linearity wasn't enough for him either. His linearization scheme didn't alter the internals, rather added in PRBS noise in anti-phase between the L and (digitally inverted) R channels and then summed them. As far as I can recall anyway, this was going back to the early 90s.
 
John Atwood was recently able to buy a tube of TI/Burr-Brown PCM-1704U-K's for $75 each. Oddly enough, the super-grade K series are more available than the lower grades. It's a good question whether or not the PCM1704's you can buy are actually *current* production - maybe TI only wakes up the line when demand builds up enough to justify a product run. (That's the way audiophile caps are made, by the way - they're simply a special, low-volume run made with different films.)

Renee's description of the 20400A lines up with everything I've heard: they're custom-trimmed PCM63's, which offer the unusual option of external trim of the LSB's. Renee, it's the lowest two bits of the 20-bit converter, isn't it?

One difference for the passive I/V conversion crowd is the PCM63 runs at twice the current of the PCM1702 and PCM1704, so a PCM63 can be used with a 100 ohm resistor without turning on the internal diode, while the PCM1702 and 1704 require 50 ohms (preferably a bit less). I do remember measuring the PCM63 and finding that yes, 100 ohms is fine, while 105 ohms you start to see the very first signs of little distortion spurs emerging from the noise floor. With the spectrum analyzers that Matt Kamna had at the time, we never saw any difference in distortion between 10, 20, 50, or 100 ohms.

Regarding the Karna amplifier ... yes, I've been occasionally tempted by direct-heated input tubes, since the already high-performance 5687/6900/7119 series is the only 5th-and-above source of harmonics in the entire amplifier. The PP 45 section is almost frightening to measure: Gary Pimm and I only saw 2nd and 3rd, with the instrumentation noise level at something like -130 or -140 dB. Gary's test setup is pretty exotic, and uses custom-built test fixtures to achieve the astonishing dynamic range. The PP 300B's are a little dirtier, with the 5th way down at something like -80 dB, and the 6th and 7th something like -90 to -110 dB. When I say the Class A PP 45 and 300B are low distortion, I'm not joking.

Old-stock 2A3's were not so great, with 5th around in the -60 to -80 dB range, but this is very vendor-specific, with a remarkable 20 dB spread between brands. To our surprise, varying the operating current only made a 6 dB change in the high-order distortion spectrum, and sample variation was in the same range as well. It was the *vendor* that made the difference. Obviously the difference was in internal filament and grid structure, and most likely the result of the jig used in manufacturing. (I don't have the nerve to try cryo-treating a vacuum-tube, since I'm concerned about damage to the glass-metal seal over time.)

All of the indirect-heated triodes were another step down, with the best we could find somewhere around -60 to -70 dB for the 5th-and-up harmonics. We tried a *lot* of these things, and once again, it was the vendor that really mattered.

Note that the levels of 2nd and 3rd were all pretty similar, falling in a 6 dB range, with high-order harmonics what set them apart. The test circuit was the same as a Karna stage: transformer in, transformer out, all Class A PP, with cathodes bypassed. This was the lowest distortion configuration of any. We tried a current-source cathode (forcing the tubes into exact match), and high-order harmonics skyrocketed. John Atwood, using his Audio Precision setup, has confirmed the same result: a high-impedance shared-cathode circuit lowers 2nd harmonic (which is nearly inaudible) at the expense of high-order harmonics; peak current delivery also goes down by a factor of 2 or more.

If you're really clever, John found an interesting result: there is an optimum shared-cathode dynamic impedance (which is not the same as the cathode resistor) for the lowest possible upper-harmonic distortion. In practice, for a circuit with a shared cathode resistor and a cap bypass, you insert a pot in series with the bypass cap, and adjust the pot while looking at the spectrum analyzer, and look at high-order distortion at the -3 dB and -30 dB modulation levels.

At the 2004 European Triode Festival, the audience got to hear what addition of 2nd harmonic, 3rd harmonic, and 2+3rd harmonic sounded like on a (very) high-quality system with musical selections (the distortion was generated by computer algorithm, as I recall). I was surprised just how hard it is to hear 2nd harmonic: for me, the threshold level was around 1% or so, and sounded kind of like an old jukebox, a warm syrupy sound, like you'd expect. 3rd by itself was audible around 0.3%, and sounded kind of rough, like a trumpet blat, but not too awful. The combination of 2nd and 3rd in a 3:1 ratio sounded surprisingly like the undistorted original, but a bit more rich, more dense, a little thicker, if you will. So these two harmonics, by themselves, are pretty innocuous, and the higher-order harmonics are much more unnatural and "electronic" sounding.

When designers go to extreme lengths to eliminate 2nd harmonics, they are barking up the wrong tree, since it is so hard to hear. Reducing harmonics from 5th through 9th and higher, though, is really worthwhile, since it has a big impact on IM distortion.
 
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^ So sorry, but triode amps are like trying to stomach Limburger cheese. If you love it great, if you don't, "forgetiboutit!"
 
Going back to the original question about the Karna amplifier, what it is really missing is a high-quality DC source for the filament supply. I've tried 3-pin regulators for this application, and they sound unbelievably bad; they make the amplifier sound like a cheap home-theater receiver ... grainy, flat, 2-dimensional, you name it. Obviously some of the audio signal is flowing through the electronics of the cheap opamp in the 3-pin regulator, and it sounds like a 741 opamp, or worse.

My friends over in Germany report good results from a very old-school approach of floating LCLC filtering of a Shottky or HEXFRED bridge ... no regulation per se, but then again, it isn't really required. What is required is electrostatic screening of the secondary of the filament transformer, since the filament circuit both floats and carries audio currents.

One little lesson I learned is that filament circuits are *extremely* sensitive to quality degradation: if you find that XYZ brand direct-heated-triode amp has a noticeable "solid-state" coloration, look there first. Clip out the filament junk and revert to AC heating, and you might be surprised at the magnitude of the improvement.

No, I don't think AC heating has some sort of magic to it, since I've heard SET amplifiers with DC heating that sounded really good ... most noticeably, the ones made by my German friends, that I heard at the 2004 ETF.
 
^ So sorry, but triode amps are like trying to stomach Limburger cheese. If you love it great, if you don't, "forgetiboutit!"

I suspect you've never heard a transformer-coupled PP DHT amplifier; most audiophiles haven't. They're kind of rare in North America.

95% of the commercial SET amplifiers are awful, really gruesome sounding. Back when Karna and I were listening to a lot of different amplifiers on the Ariels (we must have heard 30 or more, all kinds of topologies, including the Crown Macro Reference), we heard a well-known and popular SET amp that sounded like an old transistor radio that had been left in the sun too long ... much worse than any car stereo I'd ever heard. We both looked at each other and just laughed. No, it did not get a review from me, although other PF reviewers thought it sounded great. No accounting for taste.

One thing about SETs is the remarkable scope for getting things wrong: no feedback to hide the mistakes, so you hear the screwups in all their glory. They can sound really shrill, really flabby, grainy, grossly distorted, like an unrestored jukebox, sometimes all of the above.

Here's a recipe for really bad SET amp: solid-state rectifier bridge, cheap guitar-amp style RC filtering for the main B+ supply, feeble driver tube running at 3mA or less, cap-coupling with Solen or electrolytic caps, cheap 3-pin regulator for the DHT filament supply, and Chinese output transformers. Plenty of overpriced SETs with fancy cases are just like this on the inside.
 
IME, and from what I recall of various papers through the years, we are more sensitive to 3HD and then higher-order terms than to 2HD.

My old tube preamp (ARC SP3a1) and a few others I measured had vanishingly low distortion, unlike most tube amplifiers I measured. Although, IME/IMO, the biggest failing of tube amps is not their higher distortion (which was still plenty low), but that they are very sensitive to the speaker's impedance due to their high output impedance.

I found the sound of my old SP3a1 improved with regulated filaments, but although 60 Hz noise decreased wideband noise actually rose slightly (less hum, more hiss). I had to incorporate much more decoupling than I had anticipated. I think what you hear is not signal through the regulator, which seems highly unlikely, but modulation of the signal by the filament supply and vice-versa. That sad, I have very little experience with SET circuits. At least partly because my favorite speakers (Magnepan) are relatively power-hungry...

Only 3 mA drive for a class-A output stage sounds pretty low...
 
I suspect you've never heard a transformer-coupled PP DHT amplifier; most audiophiles haven't. They're kind of rare in North America.

95% of the commercial SET amplifiers are awful, really gruesome sounding. Back when Karna and I were listening to a lot of different amplifiers on the Ariels (we must have heard 30 or more, all kinds of topologies, including the Crown Macro Reference), we heard a well-known and popular SET amp that sounded like an old transistor radio that had been left in the sun too long ... much worse than any car stereo I'd ever heard. We both looked at each other and just laughed. No, it did not get a review from me, although other PF reviewers thought it sounded great. No accounting for taste.

One thing about SETs is the remarkable scope for getting things wrong: no feedback to hide the mistakes, so you hear the screwups in all their glory. They can sound really shrill, really flabby, grainy, grossly distorted, like an unrestored jukebox, sometimes all of the above.

Here's a recipe for really bad SET amp: solid-state rectifier bridge, cheap guitar-amp style RC filtering for the main B+ supply, feeble driver tube running at 3mA or less, cap-coupling with Solen or electrolytic caps, cheap 3-pin regulator for the DHT filament supply, and Chinese output transformers. Plenty of overpriced SETs with fancy cases are just like this on the inside.

Really? Why do you keep forcing your love of triodes on us common folks?;).. And never suspect what I have heard.
 
I've seen SETs that use 12AX7's for the driver tube, biased at 1mA. That guarantees slew distortion at less than 5kHz. Absolutely idiotic choice. In practice, if you care at all about slew rate or the ability of the driver to shrug off nonlinear grid impedance of the DHT, you need an absolute minimum of 10 mA, with 20 mA giving much better results. The other gotcha, routinely violated in just about every commercial SET I've seen, is *more* upper-harmonic distortion in the driver than the power tube ... as a result, most of the distortion you hear comes from the driver, which is not good design practice.

I once owned a SP3A. Nice build quality, but as I learned more about vacuum tubes, I realized it was impossible to modify thanks to the bizarre decision of ARC to run the heaters in series (not good practice), and the use of circuit boards throughout. Any mods would amount to completely re-wiring the circuit boards, which would destroy them.

The biggest problem was the use of 12AX7's for cathode followers. No!!! The 12AX7 does not run much hotter than 1mA, at the most, and the cathode-follower will not have enough current to drive the RIAA feedback loop; the preamp will slew every time there is a pop or tick on the disc, which is a good part of the time, especially with wideband MC cartridges (MCs are flat to 50 kHz and above). Since there is little current available to re-charge the caps in the RIAA loop, recovery from slew takes a long time, and distortion is very high while the circuit is recovering.

The other gotcha was 12AX7's for the output tube; again, not enough current, and every extra 100pF of cable capacitance will audibly drag down performance. Bad design. Admittedly, all ARC did was copy a Golden Age preamp and add solid-state B+ regulation, but that's still no excuse for poor design decisions about the cathode follower section. (Back in the Fifties, a 12AU7 .. still not a great tube, but cheap at the time ... was the usual choice for cathode-follower duties.)

Many of complaints about the mushy and sometimes grainy sound of vacuum tubes comes down to little more than using the wrong tube for the job, particularly a low-current tube asked to deliver more current than it can. This is bad design in the solid-state world as well; with solid-state, you get grainy and harsh sound when not enough linear current is available to drive an internal feedback loop, a following power stage, or an external load. When the same mistake happens in a vacuum-tube design (and it is very common, almost an industry norm), you get the notorious "tubey" sound and noticeable sensitivity to cables.

Raise the linear current by 10x to 20X, and it's a whole different world. Tube coloration gone. The most annoying aspects of solid-state gone. But this costs more; linear, high-current vacuum tube circuits with well-designed current-source or transformer loading are pretty expensive, and overdesigning solid-state isn't cheap either (heat sinks, more exotic parts, etc.). Unfortunately, in the high-end world, many designers are guided by little more than what the old guys did back in the Fifties (the sin of the vacuum-tube crowd), or pretty-looking results from SPICE modeling (the sin of the solid-state crowd).
 
What does all this amp talk have to do with DSD vs. PCM?

I'd suggest create a new thread for amp architecture.
 
Well, similar problems occur in I/V converters. The most common is using an audiophile-type opamp with a moderate slew rate (10 to 20V/uSec) as a transimpedance amplifier. The device has to deliver current into its own feedback loop at extremely high speeds, and cannot do so linearly. Many designers are not bothered by the slewing, thinking that slewing and lowpass filtering are similar. They are not: slewing at this location results in inaccurate Nyquist reconstruction, which demands linear lowpass filtering.

Going further, DSD and PCM have radically different ultrasonic spectra. DSD is close to spectrally shaped noise, while PCM is a collection of images. This, in turn, affects how the opamp slews with ultrasonic content.

This is a subjective assessment on my part, but I find that when slewing is avoided in the I/V section, the differences between DSD and PCM narrow. Unfortunately, this is a difficult comparison for consumers to make, since the majority of high-end PCM and DSD DACs use opamps or discrete-transistor circuits with fairly low slew rates in the active I/V section.
 
As an employee of Pacific Microsonics...

Hi Rene, having done some DSD/PCM comparisons myself, and coming down heavily on the PCM side, I was trying to understand why my findings were different from others who've also done this comparison. One of the factors that I felt was important was that I was using a PM2 @24/192 and then playing back on a NOS/filterless DAC capable of 24/192 replay. Back in post #110, I was talking about the ADC in the PM2 and my understanding that this is non-oversampling at the 4fs rate I use.

Ritter and Johnson talk about "24 bits" in their respective quotes in my post #110. From the graphs I posted, it looks like my particular PM2 has ADCs with 'only' 17-18 bits of resolution. In any event, the PM2 @24/192 sounds better than my Korg and Tascam in DSD quite easily. Could you shed any light on the two PMADC-1 modules in the PM2?

Mani
 
One difference for the passive I/V conversion crowd is the PCM63 runs at twice the current of the PCM1702 and PCM1704, so a PCM63 can be used with a 100 ohm resistor without turning on the internal diode, while the PCM1702 and 1704 require 50 ohms (preferably a bit less). I do remember measuring the PCM63 and finding that yes, 100 ohms is fine, while 105 ohms you start to see the very first signs of little distortion spurs emerging from the noise floor. With the spectrum analyzers that Matt Kamna had at the time, we never saw any difference in distortion between 10, 20, 50, or 100 ohms.

I don't know about the 63, but the 1704 takes ~160R *IF* you apply some sneaky tweaky the analyser at hand. This is about finding it's optimum THD figure and NOT implying that linear graph for it (that graph showing how THD decreases while bits decrease). So, making the graph not linear but with a lump (downwards) at the right side, THD will be better than full scale and IIRC its optimal window is in between something like -12dBS and -22dFBS (and linear there). So 1. increase the resistor but 2. attanuate (diode will stay open). Big gag.

For me and paralelling 4 I could create something like 600mV (IIRC) which was sufficient for my gain of 20 and 115dB efficient speakers. So, it worked. I played for over a year with that with great satisfaction.
In that same full year and throughout, I knew that this couldn't be any production version, especially because I sort of "forbid" to use preamps or any form of attenuation - I tried to have more output without degrading things. Whatever means I tried, the first what would happen is a not straight noise floor I ran into and which is my very foremost litte thing to strive for.

Idle Noise FFT 256K.png


... with this as an example of how I like it (only 4 times averaged and notice the FFT depth of 256K).
Having tried about all I/V means known to the planet and refusing to ever try OpAmps apart from discrete ones which also did not work out, I ended up with something carrying a slew rate of 670V/us. I know, not 1K, but better for my environment (which is about noise figures, voltages used and some more configurative stuff like differential setup etc.) I could not find. And you know what ? it sounded better. Way better.
My reasoning back at the time was that the high frequencies were eating the current, especially when seeing that not the bass improved but the highs. This is sort of related to the images behind the mirror which are there in my NOS/Arc Prediction Filtered case and although they are sufficiently down, they eat energy and by a higher frequency than normal audio band frequencies, obviously.

When you dig deeper in my explanation, you can see that it is all over unofficial "reasoning out" what is for the better and what won't work. I mean, let pass higher frequencies beyond the audio band less distorted (see slew rate) on purpose ? Well, if we don't do that then the in-band higher frequencies become muffled. A nice natural sort of filter ? no way. No way since those "un-muffled" frequencies come out the most undistorted. And so the fun : instead of seeing a roll off in the higher frequencies dedicated to NOS/Filterless, we perceive a high frequency output (for level) which is unsurpassed and of which at least I dare claim that it is unsurpassed indeed.
And let's say (and hope) that by now I should have sufficient experience in knowing the difference between "freshness" coming from unfiltered NOS hence a mountain of piled up harmonic distortion on one side, and super silk and undistorted highs with a perceived maybe over 20dB more output than any normally filtered DAC on the other. Ever heard sheer SPL from brushes on a snare ? 5 years back the whole d*mn brushes weren't even auadible in my system. And yes, brushes may be the toughest "instruments" to render well.

Peter
 
Interesting discussion relating to opamp/discrete implementation of the DAC-analogue stage because I always felt the sound quality issue was more to do with the design and implementation (both architecture and software) of the oversampling and filter in the context of delta-sigma converters assuming opamp/discrete circuits was done correctly, so it has opened my thoughts up a fair bit on the subject.

That said, quite a lot of listeners to reference Metronome Technologie digital equipment rate the sound as one of the best they have heard, and from what I remember this is using some pretty specific opamps.
I appreciate my context is PCM and ignores DSD.
So what are they doing that is different?
However I do feel and agree there is a lot of similarity between the various delta-sigma DACs as raised by quite a few in this thread, including those that are more hybrid such as ESS Technology and their Sabre 32.

Cheers
Orb
 
Hi Rene, having done some DSD/PCM comparisons myself, and coming down heavily on the PCM side, I was trying to understand why my findings were different from others who've also done this comparison. One of the factors that I felt was important was that I was using a PM2 @24/192 and then playing back on a NOS/filterless DAC capable of 24/192 replay. Back in post #110, I was talking about the ADC in the PM2 and my understanding that this is non-oversampling at the 4fs rate I use.

Ritter and Johnson talk about "24 bits" in their respective quotes in my post #110. From the graphs I posted, it looks like my particular PM2 has ADCs with 'only' 17-18 bits of resolution. In any event, the PM2 @24/192 sounds better than my Korg and Tascam in DSD quite easily. Could you shed any light on the two PMADC-1 modules in the PM2?

Mani

The ADC was an 18 bit unit purchased from Analogic in Peabody, Mass. It was a two stage flash architecture with internal sample and holds that was capable of running at full accuracy up to 200KS/sec. So run at 4x 44.1, with digital decimation it achieved 20 bit s/n for the intended RBCD mastering. For the PM2 to use the same part it was necessary to improve resolution by adding dither, to the level of several LSBs, to the input and subtracting out that same dither in the DSP.
This is a trick used by the instrumentation guys to get more for less. (Analogic used a massively dithered 8-bit part to achieve 20 bits in a medical application, albeit, for low frequencies.) What you get is the dynamic range extension equivalent to more bits, ie, resolution below the noise floor, but still with the linearity of the 18 bit part.

I was no longer with PM during the creation of the Model 2, but with the Model 1, each ADC was measured and tweaked if it didn't perform at the 20 bit level by adjusting the LSB current source resistors. The part was otherwise heavily modified with faster opamps and sample/hold. I saw many units that performed at the 20+ bit level for linearity after tweaking.

After the Company was sold, HDCD going to Microsoft, the last M2s were built by Euphonics. I don't know if they had the skill or inclination to make mods to the ADC. The dither ensured at least that the resolution extended to 20 bits, if not the linearity.
 
Renee's description of the 20400A lines up with everything I've heard: they're custom-trimmed PCM63's, which offer the unusual option of external trim of the LSB's. Renee, it's the lowest two bits of the 20-bit converter, isn't it?

The UltraAnalog 20400 was actually an 8 bit DAC for the MSBs and a 12 bit CMOS part for the LSBs tied together using a state machine algorithm to unsure linearity at the juncture. PM used personally tweaked parts that were further modified for use in the Models 1 and 2.

One difference for the passive I/V conversion crowd is the PCM63 runs at twice the current of the PCM1702 and PCM1704, so a PCM63 can be used with a 100 ohm resistor without turning on the internal diode, while the PCM1702 and 1704 require 50 ohms (preferably a bit less).

Using a resistor for I/V conversion is always risky (linearity-wise) with current output parts, but I know many have tried and liked it. As Lynn says, you want a part with the highest full scale current output so you can get enough voltage with the lowest value resistor and still get decent noise performance (why Peter uses several DACs in parallel.) I haven't used these parts since the mid-nineties, so don't recall the particulars, except that they were not as linear as the Ultra parts by about 10-20dB. To qualify 20400s we looked at distortion as we dropped the level through a 20dB range. As you go down in small increments, it's surprising how much crap pops up on the FFT at specific input levels.
 
As you go down in small increments, it's surprising how much crap pops up on the FFT at specific input levels.

And then to know I created a "one page" post about this today, but am reluctant to post it (say too much "classified" TI data in there).
But true, and all about those not-so-the-same resistors in there. Well, sometimes ...

Peter
 

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