300b lovers

I use 1/2 of 6sn7 in the input stage. The driver stage is the 6f6 tube drives the 300b though interstage transformer. The coupling between 6sn7 and 6f6 is RC with Vcup Cutf capacitors.

@alexberger The 6SN7 can support fairly high current which is why it can make a good driver tube if used properly. It can also make an excellent voltage amplifier (1st stage of gain) since it is quite linear if used correctly.

SETs usually do not need much gain since the speaker used with the amplifier should be high sensitivity (if a 300b power tube, +100dB is a good value). The 6SN7 will allow plenty of gain for this. One section can be used as the voltage amplifier and the other section the driver.

You should be able to get plenty of bandwidth using a 6SN7 and an interstage transformer! The issue will come down to the quality of the transformer itself.

@alexberger 

your choice of 6sn7 as driving tube is not good (high output Z, low max plate current...)

Now, your choice 1 of implementation is the better one, only your concern is invalid.

High output Z does not automatically leads to narrow bandwidth.  Bad Interstage transformer causes it.  High Z tube can't drive 300B into deep A2 so you never get 8W  class A output from 300B that way. However, you can use Western electric's classic Ultra path design pattern, combining a CCS with a 0 DC current nickel 1:0.75 interstage transformer and achieve pretty good result if you bi-amp the speaker.

I have a question about the first 6sn7 stage in my 300B SET. What is the best way to make load and coupling?
1. The drawback of the interstage transformers is a narrow bandwidth in combination with 6sn7 that has high internal impedance. But low distortions.
2. RC - coupling has wide bandwidth but higher distortions.
3. LC - coupling looks good, but what is the optimal inductance for 6sn7? The issue can be not deep enough low frequency bandwidth and high LC resonance Q.
4. What are the drawbacks of SRPP with C or direct coupling? What is for and against it?

@lynn_olson @donsachs @atmasphere - Thanks for your responses. My system is bi-amped with a SS amp driving the lower three octaves, so I'm primarily concerned about upper bass on up. I'm pretty happy with my current DIY 300B PSET monoblocks, but I like to tinker.

I've primarily been building electronics based on other's designs (at least at the module level), but I've recently been trying my own designs (heavily influenced by others). I think I want to build a PP amp as my next amp, using either 300Bs or 2A3s. My main speakers are pretty sensitive (98db/w line arrays) so I don't need a ton of power. I'm still thinking about what approach I want to take. 

Since he is no longer with us to defend his design, what do you think are the positive attributes of a differential output stage in a tube power amp?

@jaytor Since Lynn isn’t going there I’ll take this one. The advantage is the differential effect reduces distortion in the output section and makes the output section easier to drive since it will have a bit more gain.

There are a lot of differential output sections in well known tube amps- such as the Dynaco ST-70. What makes it differential is the use of a common cathode resistor. A Constant Current Source (CCS) can help performance but isn’t needed for the gain stage (whether an output section or not) to be considered differential.

Despite Lynn’s remonstrations, if designed properly a CCS in the output section of an amplifier will not limit current right up to the full power of the amplifier; in fact if the output section isn’t differential, using a pair of cathode resistors rather than a common one, the output power is unchanged or even reduced. I’ve seen applications where the use of the CCS actually increased the output power by a few Watts since the distortion was held in check to a higher power level.

What might not be obvious WRT an output section is you can set up the cathode circuit regulator to sense B+ variation and adjust the cathode voltage in response, which reduces distortion and increases tube life. This eliminates the benefit of a regulated B+ which would otherwise be a hefty lift in terms of execution and cost. IMO Lynn is missing a bet on this one and leaving performance on the table.

Yes we all have different tastes.  I would say the Lynn and I overlap considerably.  When we discussed our favourite rooms at the PAF we were in general agreement.  I stand by what I said a million pages back in this thread.  If I were to magically create a straight wire with gain that could drive any speaker load with infinite power and had no sonic signature at all, my guess is that half the people wouldn't like it.   Heck, maybe I wouldn't like it.   If I were to characterize what I favour it would be lush, not mush, an ethereal high end where cymbals and triangles hang in the air and decay, and tons of detail without brightness.  I hate artificial, in-your-face detail created by bright sounding systems.   I am an imaging freak as well.  If I build an amp or preamp and the sound stage doesn't extend at least a few feet outside the speaker boundary and the vocal doesn't appear at the floor to ceiling interface, then something is wrong.   I was a little dismayed at the show when our system could give great depth, but we had a shoe box of a room and the sound stage was constricted in height and width by the room boundaries.  Same system in my room threw the sound stage I just described.  I voice things with acoustic instruments and vocals.  If you can get a piano right you can pretty much play anything else with correct tonality.   If you can follow individual voices in a choir or instruments in an ensemble of some type, then your system can handle complex passages without breaking down.

Oh, and I love a good horn system.  Not the cheap ones that are popular, built with junk parts and cabinets, but a good horn system.  

So that is what I like.  Lynn and I agree more than we disagree!

There was a funny incident a few years back at the Dallas Audio Show. Back then, it was a little bitty thing, just a few exhibitors, but very much a home-town thing where everyone knew other. New to me, of course, as a former West Coast guy fairly new to Colorado. Never been to Texas before.

I wander aimlessly down the halls, no real goal in mind, looking for interesting tube gear. I walked into one room, and whoa, that’s Nelson Pass! Now people joke about me being Mr. Natural, but Nelson really looks the part. You can’t miss him. Me, my only trademark at a show are the Hawaiian shirts I like to wear.

Nelson had actually built an open-baffle speaker around a Lowther and a 12" guitar speaker called the Tone Tubby that I had written about some time ago. Well, that’s different, but why not? As I turned towards the door, Nelson blocked the exit.  how do I get in these situations? Me and my big mouth.

It turned out the two drivers were bi-amped with a simple low-level crossover. Oh, now I get it. Four knobs, two for level, two for the crossover frequency. Nelson wanted me to tune the thing ... by ear.

Now I really want to escape, but Nelson is still in the way. Fine, anything to get out. Twiddle, twiddle. Too little bass. Mo’ bass, man. Turn that knob up. A bit less Lowther, but not too dull. Mess with the crossover overlap some, so that mellow hemp cone transitions into the characteristic hard paper Lowther cone. A few minutes later, sounds OK, as good as I can get it right now. (Did not sound OK when I walked in.)

Escape permitted. Afterward, Nelson allowed as to how he saw that article I wrote about the charms of the Tone Tubby and wanted to build a simple open baffle around it, with a Lowther on top. So he figured if anyone could tune it on the fly, it would be me. Well, he had me there, but I allowed that he might have different preferences than I did, so feel free to mess with the knobs, although he might want to mark the current positions before changing anything.

These weird things happen to me at shows. That’s how I met Nelson Pass.

I should mention that Allen Wright liked a very different sound than I do; he liked fast, snappy, and what sounded to me like thin bass. I like a big, lush, spectacular, CinemaScope sound, the sound I heard in 70mm theaters when I was growing up. (Which had Altec Voice of the Theater speakers behind the screen, along with Altec amplifiers.)

The same applies to my brief encounters with Nelson Pass. He likes it a lot thinner than I do, but with a different tuning than Allen Wright. Kind of hard to describe, actually, since this was all a long time ago. Allen liked the sound he was getting, and he liked his own amp, even at that meeting all those years ago. What I thought was a disaster seemed OK to him. In all honesty, it was a split decision.

I mean, I didn’t like it, nor did Gary Pimm, but we were on a different wavelength than Allen Wright. His designs, like mine, are tuned to his own tastes, and we found out they were surprisingly different. Similarly, I was surprised at Nelson Pass’ tunings, very different than my own.

As it is, Don and I have a bit different preferences, but at least we are still on the same planet, so we get along. From what I heard of Allen’s designs, no way, they are too different, no good way to reconcile the two approaches. But he was a really fun houseguest, and Gary Pimm and I had great discussions with him about everything under the sun.

I miss him very much. He was really funny and one sassy dude with total disrespect for the high and mighty poo-bahs in the industry, which I very much shared.

Ouch.

None.

As you might imagine, Allen was pretty shocked at the direct comparison, since his amp had much more powerful tubes than mine, which had generic Sovtek 300B’s, good and tough, but nowhere in the same league as Vaic’s finest. I mean, a quartet of top-of-the-line 300B’s ain’t cheap, so I never went down that road.

And Allen had just given a presentation at the VSAC, only hours before, on the power of this secret circuit, which he did not fully reveal. It was a very large current source with heat sinks and all. Yes, he could have cranked up the current even more, but the heat sinks and power transistors set an upper limit on the current. It was already close to max output.

He expected that I, an old Tek hand, would be thrilled with Tek-scope type circuit. But I disappointed him. Driving deflection plates (at very high speed) on a CRT is one thing, driving a loudspeaker is quite another. And I’d been designing speakers for Audionics several years before joining Tektronix in 1979.

Scopes are about speed, and the load is a very well-defined capacitance. Cascode differential circuits are the right answer for that problem ... they’re very fast, ideally suited for square waves, and linear enough for the purpose.

Speakers are orders of magnitude slower and are inherently vile loads. The best speakers are the worst loads ... the ones that have near-resistive loads are planar-magnetics with very low BL product (which is magnetic coupling). As you raise BL product, efficiency goes up, they get snappier sounding as the coupling gets better, and ... they also get more reactive, for the simple reason the amplifier is in more intimate contact with the big, sloppy, electromechanical system. Few amplifier designers are aware of this ugly reality. They keep hoping for speakers that can never exist.

The worst thing speakers do is insert speaker colorations (through back-EMFs) into the feedback loop, where they do not belong. Feedback is great at correcting amplifier nonlinearities ... it’s fast and responds in microseconds, just what you want. Speakers have inherent high-Q resonances that are an inescapable part of an electromechanical device. The better the magnetic coupling, the worse it is for the amplifier, which has dirty spurious currents injected into the output node by the speaker.

My approach is to brickwall-isolate these back-EMFs to the final output stage, and not expose the rest of the amplifier to them. I think of the speaker load like attaching a vacuum cleaner motor to the output section ... a source of noise and garbage, nothing good about it. The amp has to ignore this racket and continue to do its job. Feedback amps can get into trouble when the error voltages get very large; this can saturate the input section, and induce additional distortion.

In a more conventional application, like a long-tail Mullard phase splitter, differential circuits have a subtle imbalance that is not obvious at first glance. On the top, or front, side of the circuit, there is the expected Miller capacitance, as per expectation. This is the inverting side ... grid goes down, plate goes up, just like you expect.

The non-inverting side can be drawn (and is better understood) as a cathode follower driving a grounded-grid stage. Rotate the other tube by ninety degrees and it becomes more obvious. This side of the circuit has very little Miller capacitance, making it ten to twenty times faster than the other side. The beautiful symmetry falls apart at (very) high frequencies. As mentioned earlier, it can never enter Class AB drive when one side cuts off, although this is not a problem if the diff-pair is not used as a driver. In a scope, you see clever bootstrap circuits and cascodes to give that extra push at high frequencies.

This is Nelson Pass’ speciality; high speed cascode differential circuits. If that’s your thing, he has an amp or preamp just for you. If you’re using transistors, this is an attractive path.

@lynn_olson - First, let me state that I've really enjoyed your, Don's and Ralph's discussion of amplifier design and tradeoffs.

I've been intrigued by Alan Wright's designs (I'm currently building a line stage preamp inspired by the RTP3D). You've done a good job explaining the downsides of a differential output stage. Alan was a big proponent of this approach so he obviously felt it had advantages. Since he is no longer with us to defend his design, what do you think are the positive attributes of a differential output stage in a tube power amp? 

I should add I am completing a large-format 2-way speaker this summer, a collaboration with Thom Mackris of Galibier Designs, and an entirely separate project from Don Sachs and the Spatial Audio team. It’s a culmination of the extremely long "Beyond the Ariel" thread over on DIYaudio.com, and the first version was built by Gary Dahl, of Silverdale, Washington.

The woofer is an Alnico-magnet 416 (15" midbass) from Great Plains Audio, the successor to Altec Lansing, using Altec staff and tooling. It’s in a low-diffraction (4" radius curved edge) 4.2 cubic foot closed box. My version will have Bubinga (African rosewood) veneer on all sides.

The high frequencies are from an Athos Audio Yuichi A290 wood horn, with a to-be-determined 1.4" exit monitor-class compression driver. Crossover will be around 700 Hz, most likely Altec-style 2nd-order. The RCF 850 and 18Sound drivers are candidates. I also have a pair of Altec/GPA 288’s in house as fallbacks.

Efficiency will be a true T/S value of 97 dB/meter/watt. With a 27 watt/channel amplifier, headroom should be, in the timeless words of Rolls-Royce, "adequate". Alternatively, sufficient for a studio monitor application.

A 20-watt amplifier and 97 dB/meter loudspeaker was pretty typical for a serious high-end system in the mid-Fifties, so it’s not as weird as it sounds. It’s only weird in the modern context of 200 to 500-watt Class D amplifiers and 85 to 87 dB/meter audiophile speakers.

Most audiophiles do not realize how stupendously inefficient speakers are. By way of reference, 92 dB/watt/meter is about 1% efficient, or put another way, 100 watts of electricity is converted to one acoustic watt (which is plenty loud).

So where does the other 99% of these pricey watts go? Voice coil heating, which isn't great considering how tiny voice coils are, and how poor thermal coupling to the outside world is. First the voice coil has radiate its heat to the magnet, which is the closest thermal sink, then the warmed magnet has to transfer its heat to the inside of the enclosure.

Since the goal is to create X amount of acoustic watts, not a clumsy form of room heating, even small gains in efficiency are worthwhile, since less voice coil heating is occurring for given acoustic output.

Aside from outright failure, another problem with VC heating is copper's change in resistance with temperature. The resistance goes up with temperature, which might be acceptable, excerpt the time constant is fairly slow, on the order of several seconds, This creates a dynamic slurring which is pretty audible.

@donsachs Yes- those Joseph speakers were nice.

I can see 88-89 for smaller speakers. I have a little 5 Watt tube amp  I designed for desktop or a bedroom system and I use a pair of Fritz Carbon 6s with it, which are 88dB and I never run the amp out of gas (but I never play it that loud either).

But if they are going to be large there's no reason they should be hard to drive. I keep telling people that if you want to get the most out of your amplifier dollar investment, its best served by a speaker that is higher impedance and easier to drive, on account of the simple fact that the harder the speaker is to drive and the lower the impedance it is, the more distortion the amp is going to make. IOW a simple way to make any amp sound smoother and more detailed is to have it drive a higher load impedance (if all other things were somehow equal, which they never are...).

@atmasphere 

Yes we agree.  If you cannot drive a loudspeaker in a normal room to adequate levels with 60 watts/ch, then you really need to reconsider your speaker choice!

The least efficient speaker I ever owned was a pair of Joseph Audio RM25si Mk2, which were 89 dB and a true 8 ohm easy load.   With the 60 watt amp I could play them to FAR higher levels than I would ever listen to in a very big living room.  Now my speakers are generally in the 95-97 dB range so I could do the same with 10 watts.   The 27 watt 300b monos hardly know the speakers are connected.....

There are so many just brilliant speaker designs out there that are 89-90 dB+ and easy loads.   Why on Earth people get huge 84 dB poorly behaved speakers that require 200 watts/ch or more is beyond me.....

The previous point about Class A operation in a differential stage still holds: what happens when more than 100% of the current programmed in a current source is exceeded?

It won't if the circuit is properly designed!

The real question is what happens when the drive to the differential gain stage exceeds the range of that gain stage. The answer is one of the devices saturates while the other goes into cutoff. Picking the right amount of current in the constant current source (if there is one, differential amplifiers do not need a CCS to work... the first circuits we built employed a bipolar power supply; the cathode resistor had the entire B- Voltage dropped across it; this limited current to the same extent that any cathode resistor might in any single-ended circuit) is the key to making sure that the design isn't limited by the CCS. Instead you want it limited by other parameters- the tubes themselves, the plate load, etc. The addition of a CCS increases differential effect- thereby increasing gain and decreasing distortion, as well as improving bandwidth, assuming that the CCS does not impose a bandwidth limit.

@donsachs I get it. I was trying to point out the difference between what sounds 'louder' and actual sound pressure; as you know from playing tube amps the two are not always the same. IMO this is one of the bigger failings of SETs with zero feedback since, more than any other kind of amplifier made, they tend to sound louder than they really are due to how they make distortion.

60 Watts isn't a whole lot less to our ears than 250 Watts is due to the logarithmic nature of our ears. So as long as the 60 Watts can adequately drive the load it can do quite well. This is the same reason we didn't try to build a super high powered class D amp. It was more important to get it right than it was to make a lot of power- as it is, it makes 200 Watts into 4 Ohms (250 at clipping). If your speaker really needs more than that kind of power to really fly, its borderline criminally inefficient, since to merely double the sense of volume to the ear, you need ten times the power. To my understanding there are no 2500 Watt amplifiers that sound like music.

@atmasphere Please don't think I was implying that "tube watts" are somehow louder than "SS watts".  Just that I have heard both driving speakers and straining and I know which I would prefer to listen to when driven hard.   When I was building the 60 watt push pull Kootenay amps I had numerous customers sell 120-200 watt SS amps after getting one.  Many of them described the amp as easily playing as loud as their big SS amps on 87-88 dB speakers and sounding better to their ear.  That of course means nothing since obviously 60 watts was plenty to drive their speakers to levels they liked in their rooms, but they were shocked what a "mere" 60 watts could do.  I am not implying that my 60 watt tube amp was the world's greatest, just that a competent 60 watt tube amp can do a heck of lot more than many people think.

As an aside, I have an old customer who has the stereo prototype of the 300b mono project amps.  He hooked it up to a small pair of Maggies that is a difficult load and told me that was the first amp (27 watts/ch) that drove those speakers that well and he had tried some large SS amps before.   So it is quite amazing what a reasonably well made tube amp can actually drive.

Glad you’re enjoying it! As I might have mentioned earlier, in person or in this forum, I’ve been listening to the Karna/Blackbird amps for twenty years. It’s just what an amp sounds like to me.

But the PAF is the first time I ever heard what my slightly older design, the Raven preamp, actually sounds like, and it was quite a revelation. Ultra-fast, vivid, and of course, dead silent. Many thanks to Don and the Spatial team for taking it to the next level, rolling in a lot of good ideas of your own. This has been a very enjoyable collaboration, and I look forward to more.

I should also thank Don for suggesting the best way for the preamp and amps to work together, as a unit. This is the special XLR direct-in mode for the Blackbird, bypassing and disconnecting the input transformer. This lets the Raven output transformer do the phase-splitting chores with its matched split secondary windings, one of the charms of custom iron, made for the purpose.

Hi Guys,

I thought I’ve give an update on the raven (preamp). I’ve had it in my house for the last few days and have been putting it through the paces. My system setup is currently, the optical rendu from Sonore going into the Holo Audio May, with raven preamp and the Don Sachs 300b mono statements. All the electronics are plugged into the Puritan PSM156. Speakers are modded Spatial M4 Sapphires. The system is in a treated, dedicated room with dimensions 17W x 25L with a partial opening in the back. Overall the raven just terrific, it’s incredibly transparent it seems to just add some air and some tonal vividness. I’ve tried running the 300b amps directly from the May using AUDIRVĀNAs volume control, it’s very good but with the raven in place the soundstage grows larger, separation increases and theres more palpability to instruments. The timbre of interments just pops more. The best part is I’ve gained all of this with no loss of resolution. I’ve yet to experience a preamp that didn’t loose resolution to good digital volume control until now!!

One of the key features I was interested in is the headphone amp in the raven. And it didn’t disappoint either! Comparing it to my Burson Soloist SL MKII with Sennheiser HD650’s. The Raven is far less grainy sounding, the proverbial veil has been lifted from the music. That same wonderful vivid tone is there in spades, and instruments are more isolated sounding, especially when panning from left to right. The Burson just smears a little bit more. 

The Raven 🐦‍⬛ definitely is my favorite preamp I’ve had in. It excels with all the genres I’ve thrown at it. (I’m currently listening to Stevie Ray Vaughan and ZZ Top 🤘) I circle pop, rock, electronic, jazz, hip-hop, and singer songwriter/folk. And i don’t find myself preferring one genre over the other. It’s been terrific with all of it. 

Some standout music to listen try

Boys at school by Spellling

Simmer by Hayley Williams

Elektro Kardiogramm by Kraftwork

Twist of Rit by Lee Ritenour

Magma by Yello

Rich by Yard Act

Risky business by ZHU (if your system can do bass turn it up to 11 and smile)

Business time by flight of the Conchords (because it’s Wednesday) 

Let me tell the story about how misreading a graph almost bankrupted a company. This is a true story.

Our chief engineer, a Tektronix veteran, designed a power amplifier called the Point Zero Three, or PZ3 (hey, I didn’t name it, OK?).

A 100-watt/channel transistor amp. It measured great, sounded so-so, but had a little fault ... it blew up without warning, and worse, took out all the power and driver transistors, scorching the circuit board as it went. Take out the circuit board, replace the power transistors (all of them), and put in a new one. Repeat as necessary.

There were days when more came back than were shipped out. Obviously, this couldn’t continue. Word was getting out, and a failure rate approaching 50% is unsustainable.

Our new engineer, Bob Sickler, looked more closely at the SOA curves for the driver transistors. Bob told me these curves are intentionally hard to read, and it usually escapes notice that both voltage and current axis are log scale. Not only that, a straight load-line is assumed by most engineers, but with real loads, that line opens into an ellipse. Once that ellipse touches the no-go line, and stays there for more than 10 milliseconds, boom!

The chief engineer, despite being an old Tek hand, had missed this tiny little detail. It turned out the driver transistors were undersized by a factor of three, so we had to parallel them in a 3-stack with individual emitter resistors to have a stable amplifier. The fix worked; but we had to recall every amplifier in the field and update the circuit board with the new driver stack. It wasn’t cheap, but it stopped them coming back, and they kept working.

This incident resulted in Bob Sickler, the new guy who saved the company, getting permission to design a new amp of his own, which became the Audionics CC-2, their most successful, reliable, and best-sounding product. They sold more than 1,000 with a failure rate of less than 0.3%, the lowest in the industry. The CC-2 was their most profitable product.

All this happened because one part, a driver transistor, had poorly understood behavior in a boundary condition. Ask for too much current for too long, go past the SOA boundary, and blooey! A scorched circuit board with mostly shorted transistors, all thanks to DC coupling propagating the single point of failure through the entire output section in less than a second. Engineers love DC coupling, but it can propagate failure very, very fast, in less time than it takes to jump across the room and turn it off.

That experience is why I am wary of dismissing boundary conditions. Poorly understood boundary conditions can destroy a product, destroy consumer trust, and take down a company. All from not reading a data sheet carefully enough.

I’m always interested in boundary conditions ... what happens when the amp leaves its happy place and a surge of current or voltage is required. Does a circuit saturate and hit the wall? Does a transistor fail? Does it store charge and "stick" for a few milliseconds? How smooth is the transition in and out of the Bad Place?

I mention this because speakers are badly behaved much of the time. They store energy for tens to hundreds of milliseconds, then throw it back to the amplifier. The feedback network might, or might not, keep correcting this, but the error overshoots can be very large and can saturate an input section.

Many power amps do not accept boundary conditions gracefully. Not just the output section, but the driver as well. Driver transistors fail when SOA is exceeded by transient reactive loads (failure to accurately read a SOA graph almost bankrupted Audionics). In tube amps, drivers can’t push enough linear current into the Miller capacitance of the output tubes. The voltage-amp section of a transistor amp can’t charge the dominant-pole capacitor fast enough, resulting in slewing.

These are all boundary conditions, and they are audible not just when they reach 100% failure, but well before that, when nonlinearity just begins. The previous point about Class A operation in a differential stage still holds: what happens when more than 100% of the current programmed in a current source is exceeded?

This is a boundary condition problem. When current is exceeded, what next? What’s after that? Does anything fail? What does current clipping look like? Are there any energy storage mechanisms that result in "sticking", a well-known problem in solid-state power stages. If sticking happens, how long does it take before it gets unstuck? In milliseconds?

The approach in the Blackbird/Karna does not use current sources, nor differential stages. Each side is parallel, but in antiphase, and all phase splitting, and re-summing, is done by passive devices, which do not have slew limitations. The 1:1 interstage transformer makes sure that recovery from A2 grid-current events happen in microseconds, not milliseconds.

Overload happens in the tubes, mostly in the output section, and the overload condition is not affected by local or global feedback, so the overall boundary characteristic is that of a (very) fast-recovery limiter/compressor. There is no hard boundary between Class A, where it remains most of the time, and A2, AB, or AB2, depending on current or voltage demand.

During the development of the Karna amp, I was in a kind of perverse mood, so I was curious just how much abuse the circuit, and the tubes, could take. I set the oscillator level so the scope display was just below clipping, around 20 watts, and the 8-ohm test load was nice and warm. I increased the drive frequency beyond 20 kHz, and as transformer gain started to fall off beyond 50 kHz, I just increased the input level to keep the output at a steady, undistorted 20 watts. Because why not?

I finally lost my nerve at 500 kHz. The scope display was still an undistorted 20 watts, with no sign of triangle waves or flat-topping, but playing around with an AM-band transmitter (with 500 volts inside) was asking for trouble. I wasn’t trying to kill anything, but sooner or later some part was going to fail. (If any of you customers try this stunt, yes, we will void the warranty, so don’t do this. Ever.)

Not many transistor amps would survive full power at 500 kHz. Some would, some wouldn’t. It’s an absurd test, with no relation to audio use. But it’s interesting to know the development prototype survived it. No, I would never do this to the current production model, and don’t you guys try it, either.

@lynn_olson

You might want to read this article:

The Power Paradigm

Most zero feedback tube amplifiers are Power Paradigm devices.

Regarding some of your comments in your post above:

How an output section behaves was not the topic when you brought up this bit of conversation (balanced vs differential). I never said anything about an output section. FWIW its possible to build a differential circuit so a tube can saturate when the other half is in cutoff. Its all about operating points as you rightly pointed out.

FWIW the first differential amplifiers were single pentode circuits; the grid being one input and the cathode being the other. IOW all tubes behave differentially- they amplify what is different between the cathode and grid. On this account, you can see that setting the operating point is the crucial bit which may or may not allow the tube to swing from saturation to cutoff. Drive has a lot do do with it of course. My surmise is Allen simply didn’t set his operating point correctly in your anecdote.

We were building tube differential voltage amplifiers before Allen came on the scene- we were the first worldwide to offer them to the public in a audio product meant for home use. My recommendation is to spend more time working with them and see if you might arrive at a different conclusion.

The above, in a nutshell, is why tube amps behave very differently at clipping than SS amps. It is why, with the right speaker of course, that a 60 watt tube amp can sound like it has the drive of a 200 watt SS amp,

In case there’s any question about why this might seem so, its how the tube output section makes distortion at clipping. A zero feedback tube output section has a very gentle clipping character; at early onset you don’t hear the amp breaking up at all, but the distortion has skyrocketed and the higher ordered harmonics cause the amp to sound louder than it really is, despite no obvious breakup. Its an illusion.

It has led to the myth that tube power is more robust than transistor power. But in simple terms a Watt is a Watt; but how distortion interacts with our ears is a different thing altogether. A sound pressure meter will reveal the truth of the situation easily enough.

The above, in a nutshell, is why tube amps behave very differently at clipping than SS amps. It is why, with the right speaker of course, that a 60 watt tube amp can sound like it has the drive of a 200 watt SS amp, and especially why these 300b monos with a mere 27 watts each can sound like a 200 watt SS amp. Those of you who heard them at the show when the system was cranked up could hear their drive capability on an open baffle speaker of approx. 88-89 dB efficiency with a stable 4 ohm that is very well behaved. You cannot clip the amps at any sane volume level, and they can deliver large amounts of instantaneous current, while maintaining their sound quality.

There are certainly speaker designs that require 200 watts of SS amp power to wake them up. These amps are not for those speakers. But any reasonably "tube amp friendly" speaker is no problem.   If you have a speaker that presents a difficult load for the amp, then certainly one of the class D amps may be a very good choice for you.  

Quick recap: actually, vacuum tubes are far from saturation when set to normal bias points. Look at a 300B, or any other power tube. Normal quiescent bias is set between 60 and 85 mA, if Class A operation is desired. If Class AB is desired, 35 to 40 mA is more typical. With 400 volts from cathode (or filament), that's a steady-state plate dissipation between 14 and 34 watts, well within the 40-watt rating.

But that's nowhere close to the peak current emission of the cathode. I've measured 250 mA from a generic 300B, and the exotic European 300B's can slam out nearly 500 mA (transient). The only time I've ever seen a 300B current-limit around 80 mA were some particularly weak Chinese tubes from the mid-Eighties ... they sounded and measured pretty bad, and were near-defective. Other vacuum tubes are similar; the recommended quiescent currents are set by plate dissipation limits, not cathode emission maximums (which are left unspecified). Transistors will melt the internal copper links, but damaging the cathode in a vacuum tube is really hard to do unless the tube is operated with no B+ present.

It's transistors that have Safe Operating Area (SOA) curves that are log-log in both current and voltage (with an additional time dimension), not tubes. The current saturation mechanisms are totally different and have nothing in common.

Unlike transistors, vacuum tubes have very large areas of peak current emission that are left untapped by most circuits. Of course, plate heating goes up when these areas are explored, but unlike transistors, tubes do not fail in milliseconds (this is shown in the SOA curves of transistors, and must be respected). It takes sustained abuse, over many seconds, before mechanical deformation dooms the plate.

I think Ralph will agree that Class AB operation is not "false". In Class AB, one device cuts off (goes to infinite impedance and conducts no current) while the opposing device goes to a large multiple of the quiescent current. In conventional Class AB transistor amps, the idling current is a tiny fraction of the peak current, and in Class AB tube amps, it's still a small fraction.

Let's look at what happens in pure differential circuit, either tube or transistor, with a current source setting the quiescent current. This circuit must always operate in Class A. Unless something fails, the current source will always deliver the programmed current ... that is a hard limit that cannot be exceeded under any condition.

The late Allen Wright actually built a PP 300B power amplifier that had a current source under the pair of VV52B's (massive Czech power tubes). He stayed at my house during one of the VSAC shows, and we compared his amp to my early version of the Karna (which has bypassed cathodes and can operate in Class A, Class A2, or Class AB, or even Class AB2, depending on current demand). Allen's output stage was true differential, and true Class A, with a powerful solid-state current source running around 160 mA (if memory serves ... this was in 2003 or so).

The two amps sounded completely different. That's when Allen, and I, realized that differential, and balanced, are not in fact the same. This is a common illusion, a hangover from the Fifties. The question is what happens when one device cuts off.

When this happens in a current-sourced differential circuit, the "ON" device can never pass more than the total current programmed in the current source (by definition). That's a hard limit. It is a brick wall. The circuit, as a whole, will always pass whatever the current source is programmed to do ... no more, no less, always the same. This is why this circuit is seen in the Mullard topology as a low-power, medium-voltage phase splitter. Allen, as a big fan of differential circuits in Tek scopes, took it all the way and used it in a power stage.

This is quite different than a Class AB, or conventional Class A, power stage. Whether cathode or fixed-bias, current flow through the output pair is dynamic. IF (a very big if here) the output tubes were distortionless, perfectly matched, AND never voltage-clipped or driven into Class AB, yes, it would behave the same as a current-sourced pure differential stage. Only then are they the same.

But we don't live in a world of Platonic ideals. Tubes are not actually the same as the tube models, they are not perfectly assembled in perfect factories by robots, loudspeakers have odd ideas when they want lots of current, and bass drivers in particular are notorious for nonlinearity and very long energy storage .., all of which affects output stages.

So a power amplifier must deal with speakers as they are, not as we want them to be. So peak current excursions can be accommodated when necessary, without the amplifier grossly departing from basic design assumptions. The loudspeaker conforms to Theile/Small equations most of the time, but both Neville Theile and Richard Small warn us that these are only small-signal approximations. They are not valid once the voice coils start to move significantly. Speakers are only linear on average, not all the time.

My goal with Class A output is to synthesize a fixed output impedance that remains constant with real-world loudspeakers, which I have been designing since 1975. I know how awful speakers are. Most power amps use 20 to 50 dB of feedback to synthesize a perfect voltage source, and they do a pretty decent job of it. With zero feedback, the best I can hope for is a fixed, moderate-value equivalent resistor, about 2 ohms or so, which a low-Q vented or closed box speaker can deal with. And an output stage that does not have a hard current limit, but soft-clips in both voltage and current, without requiring protection circuits.

A differential circuit has a current source or high-value resistor in the common cathode (or emitter) circuit, which is why they are called "long-tailed pair" in the literature. This forces differential operation, but has a limitation because the two tubes (or transistors) are effectively in series. If one device cuts off (impedance goes to infinity), then the other device is hard-limited to 2X the quiescent current. It can never go further, because the long-tail or current source hard-limits total current to both devices.

This statement is false. The devices are not in series, else Kirchhoff's Law would prevent the second device from conducting if the first were in cutoff... At any rate if one device is in cutoff, the other will be in saturation which is the limit of any device's ability to conduct!

From my perspective as an amp designer (not as a consumer or reviewer), the Sweet Spot in tube amps is from 3 watts (Class A SET) to 60 watts (Class AB PP pentode). These are all simple circuits with an emphasis on sound quality and reliability.

If you MUST have 200 watts, consider combining a modern Class D amp with a preamp like the Raven. The new Class D amps don't have the irritating and fatiguing Class AB sound, while a good tube preamp lends the sound some charm and likability.

By the way, if you are looking for value, you really should audition the Valhalla from Spatial. It took on every other high-bucks big-name tube amp at the show and came out ahead, often by a good margin. It is a seriously good amplifier at an absurdly low price.

Now, if you are looking for 100 to 200 watts of tube power ... hate to break it to you, but paralleling arrays of power pentodes does NOT improve the sound. Rule of thumb for PP tube or transistor: no more than two devices if you care about quality. Once you start paralleling arrays, there’s always just a bit of mismatch to trip you up. And that’s just DC matching, which is completely separate from matching transfer curves (AC matching). That’s a lot harder, and there’s always the issue of tubes drifting apart as they age.

The other issue with arrays of pentodes is the grid capacitance for the power tubes is multiplied, which then requires high-current cathode followers, or separate power tubes as drivers. This gets into No Fun territory as the design complexity multiplies, all just to squeeze out a few more watts.

A bit of background on cost to the consumer: if a company isn’t charging Bill of Materials (BOM) cost times four, they won’t be around very long, one or two years at most. This rule-of-thumb has been true since the Fifties (for hifi manufacturing in North America).

Not true for cars, of course, since that is a hyper-competitive, extremely price-sensitive industry that has enormous capital barriers to entry. In electronics, the Chinese are able to shave it down to two to one, most likely due to a wide range of hidden subsidies that favor exporting.

So a smart DIY’er can indeed get serious high end for medium (not low) cost, partly by pricing their labor at zero. But even a very experienced DIY’er is going to find that building a Blackbird from scratch is the same as the price of a good used car, setting aside labor and debugging time. I know several people who got stuck halfway through building a Karna and wanted many hours of my free help completing it. No, that’s not how it works. You want a Heathkit, go buy one. If you can design and build an amp from scratch, more power to you! Have fun! Be glad you don’t have to use a slide rule any more, like the bad old days.

(Yes, I have used slip-sticks. They are no fun. You’re lucky to get 2% precision, and you have to do the calculation twice because it can be off by a factor of 10 or 100.)

Back to circuits. A differential and balanced circuit are not the same. A differential circuit has a current source or high-value resistor in the common cathode (or emitter) circuit, which is why they are called "long-tailed pair" in the literature. This forces differential operation, but has a limitation because the two tubes (or transistors) are effectively in series. If one device cuts off (impedance goes to infinity), then the other device is hard-limited to 2X the quiescent current. It can never go further, because the long-tail or current source hard-limits total current to both devices.

By contrast, a balanced circuit, without a long-tail or current source, can turn on the "on" device as hard as it likes. That can be as high as 5X the quiescent current or even more. It effectively slides over into Class AB if it needs to, unlike a differential circuit, which will hard-clip if too much current is demanded. The phase splitting is done by transformers, not a long-tailed pair.

Lynn,

Could you expand further on your balanced, but not differential comment? What is your philosophy, what have you observed?

I can appreciate the desire to chase the sound quality rabbit to fruition, but it seems likely this will result in more boxes and a bit higher cost. No doubt there is a market for this. Maybe there are two markets? Two versions at different price points?

The Karna Mk II/Blackbird is bit by bit evolving towards the original Karna, but without the madness of a four chassis design. Having a separate chokes and power transformers for #1 B+, #2 B+, and the filament supply gets really heavy and awkward. Don’s monoblock approach is much more sensible, and more important, he has real-world experience of what is reliable in the field.

I design things as a thought experiment, just to see how it works out. About one design in three is a flop, and gets abandoned. You have a sound in your mind, and wonder if the real thing will sound like you imagined. Sometimes it does, sometimes it doesn’t. You never know in advance.

The Raven and Blackbird were, and are, thought experiments to explore what minimum intrinsic distortion would sound like. Zero loop feedback, and zero local feedback, with all cathodes fully bypassed. Balanced, but not differential, with passive transformers doing the summations and cancellations.

It is not SET, which require skillful arrangement of various colorations and very, very careful component selection. But it still requires careful selection of components because there is no feedback to minimize and wash out colorations. If XYZ tube has a certain sound, well, that’s what you’ll hear. If XYZ cap is imposing a coloration on the cathode circuit (which is a very sensitive circuit node), yes, it will be audible.

A big difference between the solid-state world and vacuum tubes is capacitance. Capacitance with tubes is essentially linear, aside from Miller capacitance, and even then, the delta in the capacitance is very small (no more than a percent or so). The transit time through the circuit is constant, regardless of signal. Part of making the transit time constant is passive (not active) phase inversion and summation.

Solid-state capacitance is known for varying with current and temperature, so it pays in transistor circuits to get the (nonlinear) capacitance to the lowest value possible ... if it can’t be linearized, get it as close to zero as possible. Modern transistors are much faster than previous decades, so this really helps. Build a very linear video circuit, and many problems are solved.

@cloudsessions1 Thanks for the links!

Given how much Bruno Putzeys has written about this topic it was my assumption that all of his designs conformed to his ideals. Assumptions can get you into trouble...

I'm Ok with the agree to disagree. I've met very few solid state amps that I could actually live with; hence 45+ years in business making tube amps. I do agree its less of a problem now as opposed to +20 years ago. The semiconductors needed to do the job really didn't exist in the 80s and early 90s.

@atmasphere you can see the purifi’s distortion vs frequency here: https://audioxpress.com/article/fresh-from-the-bench-a-tale-of-two-class-d-amplifiers-orchard-audio-bosc-and-purifi-audio-eigentakt-eval1 here: https://www.audiosciencereview.com/forum/index.php?threads/review-and-measurements-of-purifi-1et400a-amplifier.7984/ and here: https://www.stereophile.com/content/nad-c-298-power-amplifier-measurements

Now the Purifi is a good sounding amp with excellent employed feedback. But it does not rival the best tube gear I’ve heard let alone my $5k Valhalla. 
 

As for THD+frequency. Agree to disagree. I have not heard any properly designed SS that isn’t good in the treble whether it has a rising distortion plot or not. to me that was a problem in the 80’s and 90’s. Designers have long known the impacts of high order distortion and taken steps to reduce it in the last two decades. The Pass labs XA60.8 has some of the best treble I’ve heard. It’s sweet, articulate and smooth and it’s distortion rises over 1/2 a percent at 20k. 

As for rising THD versus frequency, I haven’t experienced a refinement of treble with linear THD across the spectrum. Properly designed SS has been overall terrific in the upper registries over the last 2 decades. Pass labs, benchmark, and purifi all have terrific top end and all of them have rising THD versus frequency.

@cloudsessions1 Just so you know, this statement is false. Most self-oscillating class D amps, such as the Purifi, do not have rising distortion with frequency. Where ever you got that your source is wrong.

Regarding this comment:

Humans are inherently bad at hearing harmonic distortion don’t take my word for it there’s many blind tests you can do online to see how much distortion it takes before you notice.

This test is probably not done with attention paid to distortion rising with frequency- and in that context your statement is correct. Most of the online stuff I've seen does not have that built-in to the software. So its not the same thing. When distortion rises with frequency, it puts emphasis on higher ordered harmonics. This is at the root of why solid state has had a reputation for being harsh and bright, and also why feedback has gotten a bad rap in high end audio (because it can mess with a tube amp in a similar fashion).

I've already described how Gain Bandwidth Product causes the rise in distortion with frequency. What I've not mentioned in this thread so far is how feedback is usually applied in amplifiers so that the feedback signal itself gets distorted before it can do its job mixing with the incoming audio signal. As a result higher ordered harmonics and intermodulations are created because the feedback node is not linear. Norman Crowhurst (a well known audio guru of the late 1950s and 1960s) wrote about this over 60 years ago, but almost nobody really did anything about it.

You can apply feedback without distorting it. That is done the way opamps do it, by mixing the feedback with the audio signal using a resistor network at the input of the amplifier, rather than inside the amplifier. Resistors are far more linear than any tube or transistor! We've employed that technique in our smaller OTLs for decades now.

#hot take, one of the least important measurements is THD. Humans are inherently bad at hearing harmonic distortion don’t take my word for it there’s many blind tests you can do online to see how much distortion it takes before you notice. it tends to be shocking how much distortion there is before you notice it (especially if it’s low order). I’ve always had trouble correlating all things I hear with measurements. Still can’t really find a measurement that tells me how black the background of a component is. It doesn’t seem to be noise floor. I’ve heard many amps that have an incredibly low noise floor that aren’t very black sounding, other amps that have quite a high noise floor and are very black sounding. Multi tone seems to loosely correlate with this but again I’ve heard components with incredibly low and linear multi tone that aren’t very black sounding (Insert class D here). 
As for rising THD versus frequency, I haven’t experienced a refinement of treble with linear THD across the spectrum. Properly designed SS has been overall terrific in the upper registries over the last 2 decades. Pass labs, benchmark, and purifi all have terrific top end and all of them have rising THD versus frequency. And all 3 of those amplifiers employee very different topologies. 

As for component cost I completely agree with Lynn. If a component is cheaper to build, why are you charging me so much?!? I have no issue if a designer thinks a cheaper part sounds superior then a more expensive implementation. But you better not charge me more for that 🤨. This is something I appreciate about Atma-sphere’s class D. Ralph fundamentally believes it sounds better than what he was putting out before but he didn’t go charge an arm and a leg for it because “it sounded better”. 

To me it’s clear most of us have a slight different preference to the sound we like. The thing that makes the Karna mkII (blackbird) so attractive to me is just how much you can change the sound depending on what tubes you roll into it. Other tube amps I’ve heard do not change nearly as much as the Karna mkII. It is spooky transparent to what’s around it. Don very much likes the Linlai WE300B, but to me they aren’t my sound a little to smoky jazz club vibe sounding. Roll something else in and it’s a completely different presentation. Last night I was rolling the 6v6s and it was shocking the difference. Rolling in the JJ’s it was that classic JJ snap and speed in the midrange with a completely unrefined top end 🤮. Definitely won’t be sticking with that tube. But anyway my point with the Karna mkII is I’m not constrained to what Don and Lynn thinks it should sound like. I get to choose what it sounds like and that’s my favorite thing about it. 
 

Thanks,

Cloud

Why buy an high-end audiophile component, with audiophile pricing, made from off-the-shelf $5 parts.

@lynn_olson

It might be because those parts work...

Tube gear costs a lot because transformers and vacuum tubes are inherently labor-intensive, and the parts are not inserted on circuit boards with pick-and-place machines. I’m one of those madmen who think zero-feedback circuits are interesting, and I like tubes. Nelson Pass is your man if you like zero-feedback JFET/bipolar transistor circuits.

If you are more sensible, read ASR reviews, ignore the comments section, ignore the single-dimension SINAD number, and look at the noise floor of the multitone IM distortion graphs. That is the true wideband IM distortion, and multitone is the most severe test of the entire circuit.

FWIW, we use surface mount parts in the module we designed for our class D amp. We assemble them to the board by hand (no machines). You use different tools for that- a different soldering station, and special reader’s glasses so you can see what you’re doing.

You missed one of the more vital measurements: distortion vs frequency. Why this is important is that it can show you if the amp is going to make more distortion (and audible, annoying distortion) than the specs would otherwise show.

Zero feedback amplifiers have a ruler flat line across the audio band in this regard. Beyond that the distortion spectra must allow the distortion to be innocuous. That’s why SETs sound they way they do.

When the amp has feedback, that’s when you can have troubles with distortion rising with frequency. This happens because the design, whether tube or solid state, has insufficient Gain Bandwidth Product (and also points to poor engineering; feedback is control theory, which is a field that is well understood elsewhere in the electronics industry). For those that do not know this term, GBP is the frequency where the gain of the circuit has fallen to a value of 1 (unity gain) and so is the highest frequency where a sine wave can be relatively undistorted. Obviously an amp with a gain of one is not useful- 25 to 30dB is more useful so a preamp can drive the amp in a conventional manner (SETs don’t need quite so much gain, but since they don’t usually use feedback they aren’t part of this discussion).

For example if the amp has a GBP of 1 MHz and we are looking for 30dB of gain (a gain of 1000) out of the design, you divide 1MHz by 1000 and you get 1KHz. That is the frequency where the feedback will fall off on a slope (starting at 6dB/octave, but as frequency is increased, falling off faster)- and the distortion will rise on a converse slope.

This is why a simple THD value can hide dirt under the carpet; the fact that distortion will be much higher at 7KHz than it is at 100Hz. Its why most solid state amps can play bass just fine, but sound bright and harsh- you’re getting more of the audible annoying kinds of distortion at higher frequencies than the specs otherwise show! This has been one of the bigger disconnects between the spec sheets and what we hear over the years and has given rise to the myth that there are things we can hear that we can’t measure and explains why amps that ’measure poorly’ can sound so good.

It is recently become possible to build solid state amps that have so much GBP (we have 20MHz in our class D) that the distortion vs frequency is a ruler flat line, just like in an SET (but of course, overall much lower distortion, so greater detail is audible since distortion can obscure detail); IOW the feedback employed in such amps is supported across the entire audio band. That is why its now possible to build solid state amps that sound for all the world like the best tube amps.

FWIW ASR does on occasion graph distortion vs frequency on their site, but its apparent to me that they don’t understand its significance: the line that exists between that graph and what the amp actually sounds like. If you have all the measurements you can know that!

This is where my good friend Lynn and I disagree:)   I have heard my Lampi Pacific in the same system as the May (which Lynn doesn't favor).  I have heard various tube based DACs, the Schiit Yggy and a few others for SS, and the Pacific is in a different universe to my ear than any of the others.  I have not heard the Bruno Putzey DAC except briefly in the Songer/Whammerdyne room.  I liked the sound, but I need to hear things in a known system.  That DAC was over $10K though.  I will bet money that in a blindfold test I will prefer the Pacific or perhaps the expensive SS DAC over the $1000 Chinese dac du jour on ASR.   Of course the rest of the system has to be totally transparent for such differences to be heard.  My 2 cents and others will disagree.  Of course the law of diminishing returns kicks in very hard somewhere about $1000-2000.  The Pacific and others of that class live and breathe in a way that even the May cannot (to my ear).

I guess the micro-rant above is about audiophile components where the case costs more than the parts in the audio circuit. It doesn’t make sense to have a $500 case housing $50 worth of parts (in the audio circuit), unless the look is the main reason to buy the product.

I didn’t mean to imply any connection between Bruno Putzey and the Shenzen group. Bruno works in Europe, and designs top-class DACs and Class D amplifiers, which are inherently complex and very hard to get right.

The Shenzen engineers have slowly but surely improved their game, and the Chinese have been quietly building complete OEM products for high-end European and American famous-name manufacturers. Many deluxe and high-end raw parts are built right there in China, so they don’t have to go far to design and build their own high-end components, Whether you love or hate ASR, they have uncovered some terrific Chinese products.

On the international scene, a lot of truly remarkable products are coming out of Eastern Europe these days. Some real talent there.

Due to package heat-dissipation limitations, most op-amps operate in Class AB. Now, they have a stupendous amount of feedback, and it takes a difficult load to excite the AB transition, but it’s still there. Speed is the friend of op-amps, so the high slew rate versions (more than 20V/uSec) often sound cleaner and smoother than the slower versions.

It is difficult to build a discrete solid-state circuit with distortion specs that exceed an integrated circuit, but it can be done, and they are often seen in the pro recording studio world. These do operate in Class A, and that is always mentioned in the sales literature.

I'm a bit surprised that ARC uses op-amps in their "Reference" series. Why not just buy Topping or SMSL and get even better performance, or if you insist on made-in-America, Benchmark, who live up to their name in performance standards?

Why buy an high-end audiophile component, with audiophile pricing, made from off-the-shelf $5 parts. What's the point? I don't see the value proposition. Now, if there are a zillion discrete transistors, and it does 1000V/uSec and delivers 500mA mA into a 300 pF load, that's insane, but still in the realm of engineering possibility.

Tube gear costs a lot because transformers and vacuum tubes are inherently labor-intensive, and the parts are not inserted on circuit boards with pick-and-place machines. I'm one of those madmen who think zero-feedback circuits are interesting, and I like tubes. Nelson Pass is your man if you like zero-feedback JFET/bipolar transistor circuits.

If you are more sensible, read ASR reviews, ignore the comments section, ignore the single-dimension SINAD number, and look at the noise floor of the multitone IM distortion graphs. That is the true wideband IM distortion, and multitone is the most severe test of the entire circuit. The Shenzen group of manufacturers have some really good engineers, and it shows in the IM distortion measurement. From what I can see, Bruno Putzey and the Shenzen guys are at the top of the game, if specs are at top of the list. They also know how to "tune" a power supply to get a subjective result.

lynn_olson

... On paper, op-amps can do an amazing job driving a cable, in practice, not so much ...

That’s a quite confounding claim. What components using op-amps do you find objectionable?

There are many excellent differentially balanced components that use op amp circuits, including all of the ARC Ref series preamps and amplifiers. Many people consider that a better approach than transformer-coupled circuits.

@atmasphere - makes sense. Thanks for the detailed explanation.

if the input of the next component is balanced and not referenced to ground (e.g. transformer coupled), I don't understand why it is necessary to decouple the output in the source component from any ground reference to achieve the full benefits of balanced connections. Can you please help me understand. Thanks.

@jaytor Part of the issue driving interconnect cables is how the signal travels in the cable. When the shield is part of that connection, its more likely to pick up noise and the actual construction of the cable (what sort of insulation it uses and so on) becomes more critical. That shield is connected to chassis ground at the input of whatever is being driven- so now you also have the possibility of a ground loop too.

So when the source is referencing ground, such as a pair of single-ended outputs, one of which is out of phase with the other, you have a problem where the ground circuit return is active in the shield of the cable. Suddenly the dielectric in the cable is playing a role that it did not when the shield was only used for shielding with no signal on it.

It is precisely this problem which is why there are 'high end audio' balanced line cables now that might cost up to $1000/foot or more (put another way, most 'balanced outputs' on 'high end audio' equipment actually references ground as if the designers were not aware of the balanced line standard)! If the connection is done properly, you won't be hearing the sort of differences between cables that might convince someone (who might have a touch of audiophile nervousa) to spend that $1000/foot.

I'm saying that an inexpensive cable can sound just as good in every way.

The proof of this is the vast number of recordings that were made in exactly this way- proper balanced outputs and inputs. Its part of why you could have 150 feet or more of interconnect cable between a microphone and the input of the tape recorder in 1958, nearly 20 years before Robert Fulton showed off his first 'high end audio interconnect' cable, yet the resulting recording just gets better and better as you improve your system's ability to winnow more information out of that recording. That can only happen if the cables used to make that recording are absolutely transparent!

Put simply, you have to dot your 'i's and cross your 't's if you want this system to work properly.

But let's look a bit closer at that balanced source that references ground. It may well be rack mounted in a relay rack and through that rack its chassis is grounded to every other bit of equipment in the rack or maybe even in the studio. Some of that equipment might be on the input side or the output side. So a ground loop could easily be introduced! 

You might think that because you're not using a 7' tall steel relay rack at home that you won't have that problem, but keep in mind that the equipment is also grounded into the wall. That's where you get in trouble: you must be sure that ground is ignored with both inputs and outputs; that ground is only used for shielding in cables and never for any kind of signal ground! If you don't do this, the benefit of balanced operation is eroded. It was designed so that exotic cables aren't needed and grounding issues are eliminated.

Think about the advantage of having cables that sound as good as the best out there price no object, but not having to pay that price- for all the interconnects in your system, you might have only a few hundred dollars invested at the most, rather than $1000s or $10,000s. And they don't go out of date or any such thing...

@atmasphere - If the input of the next component is balanced and not referenced to ground (e.g. transformer coupled), I don't understand why it is necessary to decouple the output in the source component from any ground reference to achieve the full benefits of balanced connections. Can you please help me understand. Thanks.

The idea of introducing a tube-driven class A stage to achieve "better cable drive, partly signal conditioning, scraping off RFI and noise induced in the cables" is appealing. How would you recommend I learn about this?

@lewinskih01 You might want to study how balanced lines work. Properly done, balanced lines are the best cable drive available to audio. RFI and noise are rejected due to the low impedance aspect of balanced lines (in the old days the studio line inputs were 600 Ohms; these days its more like 1-2KOhms); weak signals induced in the cable are swamped by the low impedance. In addition the input that is being driven has a high Common Mode Rejection Ratio, which is to say that signals common to both the inverted and non-inverted inputs (such as noise and RFI) get rejection.

In a true balanced line system ground is ignored to eliminate ground loops. If using tubes this is usually done using an an output transformer which can float with respect to ground. Its also possible to direct couple using a Circlotron output, for which Atma-Sphere has several patents.

If you are supporting the balanced line standards (AES48 is one of the standards; the other is the low impedance aspect) these two methods are the only ways to do it.

lynn_olson

74 posts

07-04-2023 at 02:08am 

Well, enough of the rant on DACs. Addressing the question in the post by lewinskyh01, what does a really good tube linestage bring to the table if the DAC can directly drive the power amps?

A sense of ease, dynamic impact, and sometimes more vivid tone colors. How? Partly better cable drive, partly signal conditioning, scraping off RFI and noise induced in the cables. On paper, op-amps can do an amazing job driving a cable, in practice, not so much. If the preamp passes a quality threshold, yes, it can improve the signal compared to a direct connection to a DAC. Found that out the hard way with first Amity amp.

Great! Thank you for the answer. Would building such a device, passing said quality threshold, be super expensive? This device wouldn't need volume control nor the capability to handle multiple sources. Maybe the device increases gain by a given amount and then listening level gets adjusted down through software volume control.

I agree it's an endless rabbit hole going down into the audibility of DACs vs upstream network settings vs software. And bleeding edge DACs bleed out their value soon after their peak in fame. Yet some more professional-oriented devices (such as Merging Horus/Hapi, Prism Titan, Lynx Hilo) are worth the same today as they were 7-8 years ago (nominal prices are higher due to inflation, price of Cu, etc) and still are the company's reference product. I like to stay among these, which of course are the ones capable of doing 8-ways.

My gut feeling has been there is something else good preamps achieve, and your post helps put this into more specific words. I'm not aware of any commercially-available product that does this and I'm intrigued to explore and maybe DIY. The idea of introducing a tube-driven class A stage to achieve "better cable drive, partly signal conditioning, scraping off RFI and noise induced in the cables" is appealing. How would you recommend I learn about this?

Love ’em or hate ’em, DACs have gone a long way in the last thirty years, and continue to evolve pretty quickly. The internals of the AKM and ESS converters run at 90 MHz, with stupendous processing power. It’s what makes 4K TV and digicams possible.

That kind of speed makes up for many sins, and lets the noise-shaping algorithms operate much, much better than earliest days of SACD and single-bit MASH converters running at 2.8 MHz. In a lot of ways, it makes the endless upsampling discussions on the forums moot, since the internals are upsampling everything to 90 MHz anyway. Might as well let the chip do it, rather than play games in Roon. (Although converting PCM to high-rate DSD forces the chip to use different algorithms, which will definitely sound different.)

It is a consciously retro decision to use antique Eighties-vintage Philips TDA1541A converters, or late-Nineties Burr-Brown PCM-63 or PCM-1704 converters. Those are true once-through flash converters, with no signal processing or noise-shaping involved. But the least significant bits are kind of marginal, since it took R2R to the limit of what can be done with laser trimming and ultra precise fabrication. Nowadays, speed and good algorithms are the answer.

Which leaves the current-to-voltage converter as the last domain of audio tweakery. Op-amps are way, way better than the 1979-vintage 5532/5534 from Philips/Signetics, but you still find these antiques in consumer DACs. That’s probably where tuning happens in modern DACs, since there is little left elsewhere in the design.

And if you want to "sweeten" things, do it in the power amp or speaker. Much easier to tweak. I think making records sound like ultra-quiet, ultra-precise digital, or making CDs smoothed-out and "analog", is taking away from the strengths of each medium. LPs sound like LPs, and PCM sounds like PCM.

PCM to DSD256 is fair game, though, so why not? It’s what my Marantz SA-KI SACD player does to incoming PCM (it has S/PDIF and Toslink inputs), and an interesting "alternate view" of PCM sources.

Lynn,

I appreciate your candid views about the efficacy of DACs at varying price points. I was in the Spatial room when you heaped praise on the Mola Mola dac in the Songer/Whammerdyne room, a far less expensive DAC than the Lampi DAC in your room.  I read an owner's report that a Topping DE90 SE DAC for $900 was, to his ears, pretty much the same as the sound of his DCS Bartok DAC that cost ~12X as much.  My audio pal with nice gear has been is a rabid needle-dropper and he bought this same $900 Topping DAC and now honestly admits that his fealty to only analogue music is over as what he hears with this modestly-price DAC is pretty much the same as he hearing with his $15K analogue rig.

DAC technology, top to bottom, is really fantastic these days.  One of the happiest days of my audio life was getting a SOTA Sapphire vacuum TT in the mid 80's and another very happy day was the day I sold the TT to a local guy a few years ago, no shipping required.  Once I got an Ayre QB-9 DAC in my system, it was game over for my TT rig.  

The magazine conventional wisdom would tell you that clarity and beauty is "euphonic coloration". That’s complete horse****. Euphonic colorations can’t add detail, resolution, more depth, and more in-the-room presence ... colorations can twiddle with subjective tonal balance, and usually adds mush, murk, or grain.

@lynn_olson You might want to play around with this applet:

https://www.falstad.com/fourier/

Select 'sine' and the little dots below the waveform are movable and represent harmonics.

It shows why euphonic colorations (which are only the 2nd and 3rd harmonics) can indeed add to (or subtract from) detail and 'dynamics' and alter your perception of depth and soundstage.

If you only play with the 2nd and 3rd harmonics, and also work with their phase, you see some interesting results. For example if the phase of the 3rd is out of phase with the fundamental, the waveform actually gets taller.

Harmonics define the sound of musical instruments. You can see from this little applet that distortion can bring out details of musical instruments or obscure them.

Well, enough of the rant on DACs. Addressing the question in the post by lewinskyh01, what does a really good tube linestage bring to the table if the DAC can directly drive the power amps?

A sense of ease, dynamic impact, and sometimes more vivid tone colors. How? Partly better cable drive, partly signal conditioning, scraping off RFI and noise induced in the cables. On paper, op-amps can do an amazing job driving a cable, in practice, not so much. If the preamp passes a quality threshold, yes, it can improve the signal compared to a direct connection to a DAC. Found that out the hard way with first Amity amp.

Hi, Lewinskih01!

You bring up two different approaches to system building. One is taking full advantage of modern multichannel DAC chips (8-channel is a common default size) and letting DSP do the heavy lifting. Taking it a bit further, tuning each amp for its own driver, rather than using a AV multichannel amp of marginal quality.

It depends on subjective priorities. Does the speaker need DSP to reach its full potential, and is DAC coloration small change in the overall scheme of things? Can’t say I blame you. Speaker colorations are obvious and gross, and DSP is the most direct and powerful way to attack them.

I have friends who own Altec Duplex 604’s and they don’t like it when I tell them the only way to straighten a 604 Duplex out is DSP ... no physically realizable crossover can fully correct it. Otherwise, you learn to live with the coloration, as Lowther owners do.

DAC coloration ... hoo boy, let’s jump into that rabbit hole, shall we? I feel most audiophiles can barely hear DAC coloration for modern delta-sigma designs ... and measurements are essentially perfect, far exceeding the 44.1/16 Red Book PCM specification. If a modern AKM or ESS converter with a circuit board full of op-amps is perfect for you, you can save big money, and jump on the DSP train with confidence. Do not pass GO, collect your $200, and enter the wonderful world of DSP. Amps built to taste are entirely optional.

Only a few people can hear differences between modern converters, and if you can’t, don't feel bad, you are part of the vast majority of audiophiles. Just buy a $700 Topping or S.M.S.L. and explore DSP. It’s what headphone jocks do these days. No shame in it.

Differences between DACs are weird and extremely subtle, and frankly you have to train to yourself to hear them. I can’t honestly recommend audiophiles go down this rabbit hole. It’s extremely expensive to pursue and full of deliberately confusing technobabble from slick marketers. Maybe not as bad as cables, but still pretty bad. Trust nothing when it comes to DACs, no matter how famous the name, or how glowing the review,

I was shocked and disgusted I could hear what sounded like "big" differences between my antique Monarchy DAC, with its Burr-Brown PCM-63K converters, and the latest confection from the Berkeley DAC (which any Topping will take to the cleaners these days). I also have the exaSound DACs which are ESS based.

I find DAC chasing neither fun nor enjoyable. The best are insanely expensive, and they go obsolete really fast. I might love the $13,500 Mola Mola I heard in the Songer Audio room, but the Mola Mola won’t be worth as much three to five years from now. DACs should be thought of as consumables that depreciate the moment you buy them.

Amps and speakers ... ah, now that is good value. Buy or build a good tube amp (and that certainly includes the $5000 Valhalla from Spatial) and it holds its value indefinitely. Similarly for speakers. The good ones cost more because the parts themselves cost more, and it takes serious design work to make them perform.

@alexberger 

I don't use tube rectifiers in power amps, but I do use them in preamps.  In your amp, if you wish to keep the 5u4g, I would consider using it for the input and driver section and using good quality diodes to drive a separate power supply for the 300b.   I use only regulated supplies, and I have my way of doing it that I don't wish to get into here.  But, yes, it is a good idea to have separate supplies for the input/driver section and the power tube sections.  What sort of supplies is up to  you, but a DHT will echo your power supply AND your filament supply very clearly so make sure they are very good.  

Good luck!

Don