I might as well jump into the deep end, and pick up a conversation I had with John a few months ago, when I was saying that bi- and multi-amped systems were not easy to pull off, with overall system integration being a big problem.
I was thinking about the known weak points of the Ariel - with only a pair of 5.5″ drivers, it’s not really strong in the 500 Hz and below region (due to cone excursion). Nothing wrong with 500 Hz and above - indeed, that’s the Ariel’s strong area, with very clean impulse response and low stored energy.
The classic solution is to use a big driver - 12 or 15-inch - and cross over somewhere between 200 and 500 Hz. This relieves the mid drivers of needing to move any distance, dramatically reducing IM distortion, and lets the big driver do the real air moving.
All very good - in principle. Just select a nice low distortion prosound driver, do a serious job on the box construction (not trivial for a large box) or better, try a dipole construction, which should get down to 60 Hz with too much of a struggle.
But now we have a problem. How do we design the crossover, specifically the high-pass section for the mid drivers? What’s awkward about doing this passively is that large capacitors are required, in the 40 to 100uF range. This is not good. The bigger capacitors are, the worse they sound. You are not going to find a Teflon in these sizes, although I guess some old oil caps could be scrounged up. These large caps - and inductors - are reserved for the mid driver, with the tweeter having its own independent crossover, which fortunately can use much smaller (and higher-quality) parts.
This is where the active-vs-passive debate enters the picture, when the passive parts available aren’t particularly good. One point in favor of the active approach is that a crossover this low isn’t going to need any exotic equalization, which isn’t fun to do actively. Both the big woofer and the 5.5″ midbass drivers are in the piston band, which means they are intrinsically flat. This is to be expected; the big woofer is at least an octave or two away from HF breakup, and the 5.5″ midbass drivers are several octaves away from their 60Hz free-air resonance. This makes the crossover easy - you can use either a 6dB/octave (1st-order) or a low-Q 12dB/octave crossover, without any requirement for equalization. (Not true for a high-frequency crossover, where equalization is almost always needed.)
All well and good. So let’s consider an active approach, omitting the noxious large capacitor for the midrange driver. The quality amplifier can be reserved for the mid and tweeter, and a big-bruiser solid-state amp used for the prosound woofer. I’d pick a UcD or Class T amp over a generic Class AB amplifier; the new amps are less sensitive to load variation than the Class AB amplifiers, and as Gary Pimm has found, have quite low upper-harmonic distortion once you get below 500 Hz.
This is starting to look good; we’ve gotten rid of some nasty parts, each amplifier is working in the frequency range where it is happiest, and there are nice overload benefits too, since amp clipping in one frequency range has no effect on the other amp.
But how to do the crossover? Don’t buy an off-the-shelf crossover - to be real blunt, these things are junk, even at very high price points. You certainly don’t want cheesy 25-year-old opamps (5532 family) that run in Class AB (almost all opamps do, new or old).
You also don’t want an incompetently engineered vacuum-tube crossover that runs 12AX7’s at miniscule currents and into plate or cathode loads that cause gross distortion. Unfortunately, I’ve described just about all the crossovers you can buy in the commercial market. (Yo, John, you hearing this? There’s a gap in the market!)
The all-transformer coupled amps I’ve been designing don’t lend themselves to cap-coupling, making it a bit awkward to use the power amps themselves as part of the crossover. Yes, there is the parallel-feed transformer-coupled approach, but one of the dirty little secrets of parafeed is if you make the cap too small, you get a nasty - and quite large - LF resonance. In the real world, parafeed caps are typically 2 to 5uF, not the little bitty 0.047uF Teflon we’d like to use for the highpass part of the circuit.
So a little bit of cleverness is called for here. Basically, you want to throw away the linestage you have now, and replace it with a linestage/crossover that has enough moxie to drive 5 to 10 meters of cable (no wussy 12AX7’s in the output stage, thank you).
You see why I said there are deep waters here? We started by trying to get away from those crummy 40 to 100uF caps, and now we have to design a whole damn linestage! Nothing simple, is there?
Still, it can be done. Use low-distortion tubes that have enough standing current to drive the capacitive (or inductive) elements of the circuit, and use other tubes that aren’t upset at driving 300 to 1000pF of cable capacitance. This combines elements of a phono preamp and linestage, although we don’t have to fight noise issues, thank goodness.
We can also cheat a little and use high-quality opamps with low DC offsets for the low-frequency part of the circuit, which is also where the fancy room equalization can be placed (shelf filters and parametric equalizers). In the high-frequency part of the circuit (above 500Hz) room equalization is inadvisable anyway, and if we’re designing a speaker, we’re smart enough to pick drivers that don’t need a lot of tricky equalization.
So in a round-robin way I’ve shown how the overall system interacts. A decision to avoid low-quality parts in the low-to-mid crossover has the consequent effect of requiring a new linestage/crossover element - a device that doesn’t really exist in the current market, no less.
I should also mention that I’ve been discussing a relatively easy (!) low-to-mid crossover between two drivers that are flat and don’t need complex equalization (room EQ is a separate issue). If we’re crossing between a mid/high horn and a big bad 12 to 15-inch driver, all bets are off. This is a completely different story.
Now everything is much harder. Why? We don’t have the luxury of generous overlap regions any more, like we did crossing between big and little cone drivers.
With the horn, the impedance curve tells the story. You don’t want to cross anywhere near the huge impedance peak at the bottom of the working range - the horn is trying to tell you distortion is very high in this region, and below that peak, output is falling like a stone and the output is almost entirely distortion. You do not want to put any power in this region at all.
So what we thought was a so-called 300 Hz horn turns out to be more like a 800 Hz horn. Ugh. You see, it’s only really a 300 Hz horn if we treat the way pros do - use a 24dB/octave (4th-order) crossover and equalize the hell out of it. That’s standard professional practice - even for one-night stands, the sound man shoots the system with a MLS stimulus, and uses automagic digital equalization to make it all nice and flat. Digital crossovers with brickwall slopes take care of all the power-handling issues, which is a big deal with a gig where down-time costs big money (ticket refunds, angry musicians, etc.)
But we don’t want to take this approach, do we? We’re talking either about a zillion op-amps (cheap ones, too) or just shoving all the problems into the digital domain, and hoping the vendor’s algorithms don’t suffer too badly from truncation and rounding-off errors. (You thought digital-domain EQ was distortionless? Uh, not quite.)
So returning back to the world of quality sound, we have some difficult choices with a 2 or more way horn system. There is the problem of extensive edge-of-band equalization, in order to approximate the smooth crossover slopes we want - any little bumps and ripples in the transition-between-drivers region automatically results in violent shifts in the polar pattern, a major source of listening fatigue. I know it’s traditional - going back to the Fifties - to just blow it off and use casual, out-of-the-textbook 1st or 2nd filters, but folks, that’s a recipe for crap sound.
Sometimes tradition is just that, crap. Not everything back then was so great - and filter design was an area where tremendous progress was made during the late Fifties going through to the early Seventies. Please, stop copying all of these stale old crossover designs from the bad old days. Filter design for a speaker is every bit is critical as it is for a Stereo FM tuner (think Marantz 10B here) or a NTSC or PAL color TV. You wouldn’t build a HDTV around mid-Fifties filter theory, would you? Well, it’s the same for crossover design.
This is an area where things have to measured using modern MLS test systems, and the physical realization is probably going to be fairly complex - whether it is done at the speaker level, with line-level equalization, digitally, or a combination. The woofer, at least, is going to be fairly well-behaved, since it is more or less in the piston band. It’s the horn that’s going to need serious equalization.
Since polar patterns of horns have a lot a rough spots, especially at the lower end of the working frequency range, you’re going to need to measure not just on-axis, but at 15 degrees, 30 degrees, 45 degrees, and 60 degrees off-axis, vertically and horizontally, and come up with composite curve.
I know that most horn designers blow this off - it’s one of the less charming parts of the horn tradition. Everybody loves to talk about horn profiles and exotic phase plugs, and nobody gives a damn about crossover design, rehashing tired old crossovers that are at least 30 years out of date. Seriously, Laurie Fincham of KEF came up with Target Function Design back in 1974, and horn designers have been blissfully ignorant of the progress made in direct-radiatior speaker designs. There’s no excuse for this - horns are harder to design and integrate than direct-radiators, not easier, so it’s about time the horn guys got with the program.
Yes, you have to use a computer.
Yes, you have to get a MLS-capable measuring system, and know how to use it.
But this is what it takes if you want a horn system that is anything close to flat, and has well-defined crossover slopes.
The pros, who are by far the majority of the horn users out there, just take the easy approach, slap on a digital 24 dB/octave crossover (or one with even higher slopes), ignore the rapid phase deviations in the crossover region, and equalize the overall result. That’s what you hear when you go to any movie theatre, or any modern concert. You can be sure it measures flat - it just doesn’t sound like it, though, does it?
That’s where the more sophisticated approach of equalizing each driver to the ideal response pays off. System integration is much better, since the phase relationships between the drivers are much better defined, and the audiophile knows where to EQ the room (500 Hz and below) and where to stop, and just accept the drivers as-is.
So what would a modern high-quality horn system look like, aside from the pretty wood horns and the exotic diaphragms? That’s a good question!
I would guess we’re talking about a quite complex crossover either at the speaker-driver level or an equally complex active-crossover/linestage. That offhand comment I threw out for John Atwood earlier would have to be revised to accomodate a set of plug-in EQ/crossover boards - tubes do the work of amplifying and buffering, and the plug-in board accomodates the required capacitors, inductors (if necessary), and resistors.
The alternative is the all-digital route, which pushes the quality problem onto the ADC/DAC conversion section, where again we want to avoid the usual crap op-amps and 100uF electrolyic coupling caps (almost invariably seen in pro-grade digital equalizer/crossovers). So just apply the usual analog modification techniques that you’d use in a cheapie CD player to the prosound digital gizmo - but be sure to apply it the ADC (input) end as well, where parts quality can be pretty dreadful.

Wow! A lot to chew on here! I’m glad that you have re-considered active cross-overs, since there are some nice advantages of coupling the driver directly to the amp output, especially for woofers. Also, by limiting the frequency range seen by each amp, there is less intermodulation distortion and overload.
There really is a need for a decent tube cross-over - the Marchand product is flawed by its wimpy cathode-follower design. I will put up in a separate posting the schematic for my “CFL” (Cathode Follower-Less) crossover that overcomes most of the usual tube cross-over problems. The problem with it is that calculating the component values is tricky, since the poles of the filter have to be “pre-distorted” to compensate for the limited stage gain.
Lots of stuff here - more comments to come, I hope.
Very interesting post as usual Lynn.
I just found this blog via Thom Makris’ site yesterday, and was thinking about this post last night… I wonder if much of the difficulty people report with getting subwoofers to blend with horn speakers is due to the phasey lower cutoff effects you describe? Since I am about to head down this road with a pair of Azura horns stuffed with Fostex FE206es-r drivers, this is especially thought-provoking.
The 1 meter diameter Azuras have a physical cutoff of (reports vary) 120-150Hz, and I was planning to run them full range with a separate amp for the woofers, but now I am thinking I will need to play with a high-pass filter, perhaps integrated into the preamp I am breadboarding…
Thanks!
/pRC
Hi Lynn,
I read ur reply post here at tub diy AA:
http://www.audioasylum.com/forums/tubediy/messages/97447.html
“My personal favorite these days are the Jupiter wax caps, which are ideally suited to the low-voltage, low-temperature application of speaker crossover caps. I think wax/paper caps are being made in Europe as well by a specialist manufacturer. (Jensen?)”
=I think it’s the late Steen Dueland’s penchant for natural materials in XO components (Jupiter’s counterpart on the EU side). I think Jensen consulted with Dueland before wrt the now common paper/oil copper caps.
http://www.duelundaudio.com/
On the subject of horns, here’s an interesting commercial effort for diy:
http://acoustichorn.com/index.html
http://acoustichorn.com/crossovers.html
http://acoustichorn.com/measure.html
..Conicals as opposed to Le Cleac’h (Azura)
fred
*Very good sounding low reactance high eff. designs w/out the use of impedance compensation in the Passive XO is wishful thinking(?); I think driver selection/combinations would be critical. It would be interesting to see the non-CF type Tube XO schema.
All comments much appreciated, folks. As you can see, they all lead to the inevitable next installment, the Mid-to-High Crossover. In that region, wavelengths matter (1kHz = 14 inches), since they control polar patterns, which are very much audible.
Also, the sensitivity of the ear in the 3 to 5 kHz region is 20dB (!) higher than in the 200 to 500 Hz region - sensitivity to distortion, sensitivity to freq response variations (0.5dB is audible), as well as variations in polar patterns (nulls close to the listening position).
This is where the rough response of horns is now a big deal, and needs to be carefully attended to with measures to control horn-mouth diffraction, compensations for violent impedance variations in the horn/driver combination, and watching for high distortion due to either excessive excursion and/or high pressures in the throat area (poor phase plug design, roughness in the throat area, etc.)
Umm, just finished scanning the “acoustic horn” links, there’s real problems with the crossover description. The writer plainly does not understand what minimum phase and excess phase actually mean - although my own understanding isn’t that precise either. However, I do know that in a “minimum phase” system there is a direct mathematical correlation between freq response and phase response - so if you equalize it flat (with a typical minimum-phase equalizer or tone control), the phase response as *also* corrected as well. This is true of all minimum-phase systems - the IF strip in a color TV, the stereo decoder in a FM tuner, and the response of speaker driver in the piston band.
“Excess Phase” is a departure from this strict relationship, and is easily visible on a MLS analyzer. Typically, it implies driver breakup, and the inadvisibility of trying to equalize the driver in that region, since freq will *not* correct the phase as well, but can make it worse, not better. When I see drivers enter this region, I either cross it over to reduce the energy going into it, or avoid using the driver at all.
Frankly, the comments on the response of different bass cabinets - closed-box, vented box, transmission line - are, to blunt, just flat wrong, and show a lack of understanding of Theile/Small theory, and the underlying assumptions of TL design (which are only partly controlled by T/S equations). The most serious issue with bass horns is the behaviour of the bass horn as it approaches cutoff and the HF response variations that results from folding the horn.
One thing that is not widely understood about horns is that they devolve to a simple pipe as wavelengths approach the same length as the horn length (front to back). This is described in much more mathematical detail in Earl Geddes’ book, “Audio Transducers”, where he shows the impedance ripples and variations in polar pattern as the horn approaches cutoff. Horns really only behave like theoretically ideal horns when the wavelength inside the horn is several times *shorter* than the horn-length.
Hi Lynn,
Nice article .
As you know , I built a set of the Ariel 6Cs . I’m soon intending to put together a sub kit to add a bit of bass weight to the main speakers . As you initially started this article from the standpoint of cone excursion and bass rolloff on the Ariels , can you have a think about your best guess for integration options on this ? The sub is a 12″ light-coned driver in a sealed enclosure . I guess my first approach will be to feed it at speaker-level and just integrate with the natural roll-off of the transmission-line bass , but at a later stage I hope to beef up the phono output drive, and drive the sub at line-level, which would give me the option to roll off the main amp with reduced coupling cap size, and push the subwoofer frequency up a bit . Interested in your comments on trade-offs , if you have time.
Thanks,
Mark ( aka IslandPink )
Lynn,
Great article. I’ve been a fan of your writing and your analytical approach for a few years now.
I was wondering about your comments about solid-state circuits. (I know almost nothing about amplifier design, so please take my questions in that context.)
While I know that OPS devices in solid-state power amps are not particular linear, how are the low-power devices? Is it feasible to use signal transistors in a low-feedback or zero feedback circuit to build a line stage which has some of the properties of a valve line stage? For instance, would a fully symmetric topology operating in Class A be able to deliver good linearity with low or zero feedback?
Secondly, how do you think solid-state amplifiers like the LNPA-150 do their magic? If you had to guess, what do you think they are doing? I’ve heard other respected reviewers (i.e. not mass-market hifi-mag reviewers) saying that the very best solid-state power amps sound very similar to the very best valve amps. Do you agree? If yes, how do those solid-state amps do their magic?
yours confusedly,
Shuvam
Lynn,
Another question about solid state audio circuits, if you don’t mind it.
Why are audio op-amps built to operate in Class B? I was under the impression that the _only_ reason amp designers use Class B is to reduce heat dissipation and increase reliability in the OPS devices. I was under the impression (illusion?) that for low-power audio circuits, there was _no_ reason to ever use anything other than Class A.
I was really surprised when I read in your writings that good audio op-amps are mostly Class B circuits. Why???
Going back to front, just about all audio op-amps are Class AB for a simple reason: heat dissipation. Class A dissipates four to five times as much heat for a given voltage swing and current delivery, which makes it awkward for a small-profile chip that might need to drive 600 ohms at low distortion (as the 5532 is specifically designed to do).
Back when integrated circuits had a “compensation” pin that gave access to the output circuit, you could be clever and design an external power stage that completely bypassed the output devices of the chip. But those days disappeared in the late Seventies.
Another clever trick is to have an external current-source pulldown that switches off the lateral PNP transistors in the chip, forcing the chip to use the faster NPN transistors. In practice, many chips don’t like this stunt, producing increased distortion.
Many engineers have designed “helper” circuits for audio op-amps over the years, with mixed results. Wrapping feedback around the additional external power stage degrades phase margin, making the chip less stable, increasing settling time, and degrading slew rate. With modern high-speed chips, using an external circuit can get pretty tricky, since everything is so fast to start with, and careful layout becames important.
The input and intermediate circuits of chips are fine (and run in Class A), it’s the output section where performance is well below an external Class A section. Unfortunately, replacing the output section, or bypassing it, is not trivial.
Going to the first question, yes, low-power transistors and FETs are pretty well behaved these days. Modern solid-state preamps can be pretty good, as long as the power supply design goes beyond using ten-cent 3-pin regulators. In terms of building a hybrid tube amp, yes you build a tube using a transistor or FET front end, and some folks have done so - there are commercial amps on the market that do this (I think the Kron triode amps have a solid-state input and driver stage).
But - most of the cost of an amp is the output stage and power supply. This is true of solid-state and tube amps. The rest of the circuitry, comparatively, is peanuts, what are called “popcorn” parts in the solid-state world. Inexpensive but with excellent performance.
So if 80% of the cost of the amp is the output and power supply - umm, well, why save money on the front end? If if there’s no front end at all, well, it isn’t much cheaper. The overall size, weight, and most of the parts and chassis cost is in the output section and power supply.
The only real simplification of this or that front end is assembly time, since that’s where the complexity is (it’s like a complex preamp with lots of functions). So putting things on a circuit board gets rid of a lot of wiring - but this applies to an all-tube design as much as a transistor hybrid. Not that I like circuit boards, since unwanted capacitance can be troublesome, and the quality of that capacitance is much lower than point-to-point wiring, where the dielectric is air.
That said, Gary Pimm’s very unusual MOSFET amplifier could be adapted for use as an input and driver stage for a triode amp, since it delivers a lot of voltage swing at very low distortion - which is what you want for a triode amp, where most commercial amps have quite a bit of distortion in the driver stage.
Lynn is absolutely right about the reason for class B in op-amps. I’ll add my 2 cents:
1. The process technology for analog chips has gotten much better. Companies like Burr-Brown/TI and Analog devices have bipolar processes with good, fast PNP transistors and good JFETs, too. There is no need to compromise the design to avoid using PNPs. I just wish that they would make a high-quality front-end chip - i.e. a nice low-noise JFET diff-amp input with a class-A output. I can then add my own power stage.
2. Another approach is to build the op-amp out of discrete parts, like they did in the early 1960s. A famous design is the Jensen 990 op-amp (this is not related to Jensen speakers, but was designed by Deanne Jensen of Jensen transformers). This was originally designed as a low-noise mic preamp, but was used as the output stages in all the older designs of Summit Audio, because they had really beefy drive capabilities. They ran on +/- 24V rails and could drive 75 ohms. They ran class B at this kind of load, but the idle current was quite high, so at lighter loads were essentially class A. A schematic is at http://members.psyber.com/dibsed//CIRCATS/je-990.pdf, and you can Google for more info on the 990.
Thanks, Lynn and John. You have pretty much answered all my questions. John, I was thinking of a different circuit when I thought of a discrete op-amp. I was thinking of the fully symmetric input+VAS stage that Randy Slone had shown in his book “High Power Audio Amplifier Construction Manual”. He was saying that it has superlative low distortion, excellent PSRR, and retained these even when driving low impedances at the output. And this thing ran from + and -50V rails.
The last question which refuses to go away is: how do the really good solid-state power amps do their thing? It is a crazy thought, but will some sort of regulated power supply improve the power amps behaviour? I know that regulated power supply rails are probably just fine for input and VAS stages, but will they be of use for the OPS stage too?
(Don`t ask me what`s going on, but I simply cannot type the “slash” character or the apostrophe character. I have deliberately typed back-quote characters above, just as replacements for apostrophes. Whenever I type the apostrophe, the “Find” dialog box opens up at the bottom of the screen. I am using Firefox 1.5.)
Actually, the regulated power supplies do the *most* good for a Class AB output stage, which is critically dependent on an ultra-low impedance supply that does not radiate A/B transition switch noise into the rest of the amplifier. Back when I was working at Audionics in the late Seventies, we could see crossover noise radiated into the low-level circuits merely be moving the power supply wiring around the amplifier - it was plenty obvious where the switching pulses were coming from, and that feedback had not much effect in getting rid of it.
The circuit of the LNPA-150 was startlingly similar to the old Audionics CC-2, with the big big difference it had full regulation for the entire amplifier. This means there are really *two* amplifiers in cascade, the audio amplifier, and the DC regulator, which is a full-power amplifier in its own right. And yes, regulators have a sound of their own, so a good implementation is more than just grabbing the application handbook and copying the circuit.
Most audio circuits have pretty modest power-supply rejection ratios (PSRR), so the sound of the supply comes right through. The Karna amplifier, along with the Aikido linestage, are pretty unusual in having 30dB or more of PSRR for each stage. Even then, with high PSRR’s, there’s still some residual colorations that come through, which certainly include the power-supply capacitors and any dynamic instabilities of the regulators.
More on Class AB with full regulation. This is the only way to get Class AB to operate in “textbook” mode, the mode that amp designers assume is present, but in reality with mediocre power supplies, isn’t present at all.
In “textbook” mode, the AB transition really is well-behaved, and the amp auditions like a very powerful Class A amplifier that is nicely load-resistant. With ordinary bridge-cap supplies (like 99.99% of all transistor amps), the sonics are quite load-dependent, and there’s lots of Class AB brittleness and grit. Not to open a can of worms, but that was *exactly* what the entry-level Pioneer receiver sounded like in the Summa room at the RMAF.
Anyone that’s done any debugging work on transistor amps is all too familiar with the nasty sound of a Class AB transition - and the more efficient and lower distortion in the loadspeaker, the more obvious it is. Of course, most listeners, and magazine reviewers, haven’t twiddled with a transistor amp and listened to what it sounds like as you change the bias, feedback, and power supply. Once you have, though, you don’t forget what they sound like, or how hard it is to chase out that nasty steely sound.
What surprised me with the LNPA-150 is just much the audibility - and measurement - of Class AB is reduced by a high-performance supply. Having the supply impedance be less than 10 milliohms at 10kHz (at the pins of the output transistor) makes *all* the difference to the AB transition - again, a closer approach to the “textbook” ideal of a perfect power supply, which some circuits critically rely on for a correct implementation.
Thanks again, Lynn.
I am really not competent to even discuss some of the issues you are describing here, since I’ve never reached the point where I can dabble with the design of an amp. However, there are a couple of beginner-level questions that come to mind, based on the little I have read.
Firstly, putting a regulator on the OPS of a Class B amp is like adding the regulator to the feedback loop in some sense, isn’t it? When there’s a sudden demand for a surge of current, not only does the amp have to respond (including its feedback loop) but also the regulator, in order to keep the supply rails stable. Is this doable? Will this introduce instability? Are there circuits I can look at, which do Class B OPS fed from regulated supplies? Can I just add a massively over-rated regulated supply built using multiple LM338 or some such chips, to get benefits in the amp?
Secondly, I am a bit confused about Class AB. Does Class AB really work well? I was under the impression that when people claim they’re doing Class AB, they’re basically doing Class B with a bit higher bias current. Apparently the transition from Class A to Class B is really more a problem than a help, and very hard to get right? (I’m merely repeating something I’ve read from one veteran solid-state amp designer, who I will prefer to leave unnamed here.) So, when you separated “textbook” Class AB from typical Class AB, were you referring to exactly this sort of problems in most real-life Class AB attempts?
I don’t want to argue with your assertions, I was just trying to get a better understanding of the matter.
Lynn,
> Gary Pimm’s very unusual MOSFET amplifier could be adapted for use as an input and driver stage for a triode amp, since it delivers a lot of voltage swing at very low distortion 45 > 845; and either Gordon Rankin’s Bugle or Gary’s 47 PP amp, IT or cap coupled to an 845 output stage.
(Actually if I didn’t already have really good 10k OPTs and the 845s, I’d probably go for the all DHT 300B) . . but as I do
. . on Gary’s website, which “MOSFET amplifier” were you referring to that ‘could be adapted for use as an input and driver stage’ are you referring to?
Could the All C4S loaded parafeed 45 amp http://www.pacifier.com/~gpimm/45_all_active.gif be cap coupled to an 845 output stage?
Or with possibly extra MOSFET stages, could the voltage swing be sufficient to drive an 845, maybe with some other changes?
Thanks,
Richard
beginner guitar amp…
Just added a new album for my guitar bits….. going to add a few more pics when I get a chance/ take some of my rack, amps & pedal boards:)…