RGB LEDs for power indicators (again)

Originally, I had been planning to provide a "standby" switch, i.e., a switch on the high-voltage DC power, separate from the heater power (for the tubes).  There would have been separate LEDs for "power" and "HV", and at one point I thought it would be clever to have the two LEDs be colours within an RGB LED.  However, how to power the LEDs was an issue, since 5v DC power is only available when the preamp is on...

In any case, the Internet talked me out of standby switches; instead, there will just be a "mute" switch.  So it was back to one plain power LED for each of preamp and power amp, presumably red.  I planned to power these LEDs from the heater power lines: which are not referenced to ground except through the "hum bal" pots.  So, the LEDs would be fed from primitive DC power supplies (diode, capacitor) running across the two heater lines.

But especially considering my use of tube rectifiers, it's nice to have a light that comes on when the HV is active, after the warmup period.  And if that is the blue, and heater power is the red, within an RGB LED, then the colour will start out red, then shift to purple as the amp warms up.  At power off, the blue will stay lit and gradually fade as the HV discharges.  This is a lot of good information (wouldn't have to be RGB of course, that part is just a cute gimmick; could just be separate LEDs).  But to use RGB LEDs, everything must be ground-referenced (common cathode assumed).  So the circuit from the heaters becomes two diodes, one from each heater line, to a capacitor which goes to ground; the red LED resistor feeds from the junction of diodes and cap.  The red LED brightness will vary as the hum bal is adjusted.

Vibro-Champ

(Please note, these mods are as-yet untested, just a "try this" suggestion.)

a digression: Fender Vibro-Champ

This doesn't directly concern my amp design; but I have based quite a bit of the amp design on the Fender amp circuit.  Anyway, here's a related project, taking a Fender Vibro-Champ amp, and re-purposing the 12AX7 tube from the tremolo circuit, to become a second gain stage.  This frees up two pots, which are re-purposed as the missing "mid" knob for the tone-stack, and the gain control for the second stage (the original volume knob now behaves as a master-volume).

One could use a pull-switch pot for the channel switching, but in my case, I want to re-use the same components as much as possible (i.e., the original pots), and I already have a non-original switch on the back for disabling the negative feedback; I can re-purpose this switch as the channel switch ("2nd stage").  Or, one could drill a hole in the front panel; presumably this would be the least desirable option for most people.

*** Edit: scratch that above, I'm now thinking the best plan may be to put a switch in place of the seldom-useful second input jack.
(Notice that in my own input section, which in many ways copies Fender, I have a second input jack, but the resistor ratios are different, to make a bigger difference between the two jacks.  The Fender is only a divide-by-2, i.e., -3dB.)

(If it's not obvious, these modifications turn the Champ preamp into something very close to the "F+" gain stage, in my amp design.  This project is both a way to make my Champ more versatile, and a way to prototype the "F+" circuit before I actually begin construction of the amp.)

input section (update)

The main update to this section is in the cathode circuits connected to the "voice" switch: I now have the 2.7k resistor permanently in-circuit, and the switch just combines more resistance with that, in parallel.  This is so that the cathode is never open-circuited (allowed to float) as the switch is turned.

power amps and power supplies

With the following schematics, the initial design is complete.  I.e., if you collect all the schematics posted so far (PNG images), and eliminate older versions which were superseded, then you'll have the schematic for a full guitar amp.  Of course, some values are unspecified ("TBD"), and no doubt there will be many changes and additions to come.

power amp design -- Lafayette LA-224

The previous post basically concludes the (initial) design of the preamp.  Now on to the power amp.  As mentioned earlier, I may eventually develop a range of power amp choices for this amp, but my preferred configuration (and the one I plan to build first) is (2) EL84 per channel.

I've been looking at the circuits of various guitar amps and hifi amps that use push-pull EL84 outputs, to see what I can copy.  One that I especially like, for several reasons, is the Lafayette LA-224.  I have an LA-224A which I've restored: it's our main stereo amp in the living room.  I love the sound, especially with vinyl, though I have to admit the distortion seems pretty high.  I think EL84s are better for guitar amps than for hifis!

I found a schematic online for the LA-224B (various scans of the same original image, at various internet sites); I believe the main difference between the "A" and the "B" is a change in external cosmetic appearance, but maybe there were other changes.  I notice, e.g., that the "B" apparently has dual-ganged pots for bass and treble, i.e., a single knob adjusts L & R together, for each; the "A" has concentric pot shafts so that you can adjust left and right separately (which is usually more annoyance than convenience: I approve of their change here; unfortunately, the cosmetic changes were not an improvement, the "A" looks much better IMO).

This is my re-drawing of just the power portion (one channel):

This combination of tube-rectified power supply, cathode-biased EL84s, and negative feedback, is the basic formula that I'm looking for, so I figure I may as well start from a known-working circuit design.

The Eico HF-81 was another possible template to copy, but that amp has R-C stuff in its feedback loop.  The LA-224 has a nice, simple negative feedback loop (the 7k Ohm and 200 Ohm resistors); it will be easy to add the presence and resonance controls into this circuit.  With the HF-81, it seems to already have frequency-shaping in the NFB loop, and... I dunno, I like it simple.

Really, the only subtlety in the LA-224 schematic above, is the R-C circuit in between the two 12AX7 stages.  From what I can tell, this is a lowpass filter with a very high cutoff frequency, around 2 MHz.  It reduces the gain well above the audio band, presumably to prevent oscillation.  (Maybe the R-C components in the HF-81 NFB loop accomplish this same frequency-compensation task; regardless, I like that the LA-224 doesn't have 'em there.)

relay control section

This is basically the only solid-state circuitry in the amp, other than diodes and LEDs.  Everything will fit on a small circuit board behind the six-switch "sx" panel.

The relay drive circuits use sections of the LM324s as comparators.  A given input goes "low" to turn on the relay, i.e., to enable the circuit section in question.  Inputs can be driven "low" either with the six front panel switches, or using the optional footswitch unit.  The LM324s drive 2N2222 transistors for greater current capacity; the transistors should handle up to 600mA each, more than enough to drive two parallel relay coils (as in the reverb section).

The 1M resistors produce a hysteresis band through positive feedback: i.e., if the input voltage drops to a low enough value to just barely flip the output to "high", it must then rise more than the width of the hysteresis band (perhaps about 0.5v) before the output will flip back to "low".  This prevents oscillation or noise when the input voltage is close to the threshold voltage; the threshold shifts slightly, making the output "reluctant" to change.

This control circuit is completely isolated from the audio portion of the amp (note the use of a different ground symbol).  Power comes from a small plug-in DC power supply (aka "wall wart"), which plugs into an AC outlet provided for this purpose, inside the amp.  This outlet is switched by the "preamp" power switch.

Originally, I thought I could use the stereo effects return jacks as power amp inputs; doing this would have required some tricky wiring in the bypass relay circuits, so that the jacks would function properly even when the preamp power is off.  However, I have since relocated the effects returns such that they can't double as power amp inputs, anyway.  So, no tricky wiring, all six bypass circuits operate the same way.  And there will need to be four additional jacks provided on the back panel: stereo pairs of "pre out" and "pwr in".

"type F+" gain section

This circuit starts with the Fender AB763 circuit, as used in many of the Fender amps from the "Silverface" era.  In front of the first triode section, which would be the guitar input on an actual Fender amp, I place the "gain" pot; this allows the gain to be trimmed down, because this section will normally be fed from the input section, which also contains a similar gain stage (i.e., more gain stages than the actual AB763).  Alternatively, this arrangement allows more gain/overdrive to be obtained than an unassisted Fender amp could normally produce.

But for even more gain, I provide two more triode stages with a second tube, accessible with the "pull 2-stage" switch.  The first of these triodes is set up as a cold-clipper, similar to Marshall.  There is a "pull alt voice" switch, which increases the cold-clipper cathode resistor, and also changes the R-C circuitry between the two triode sections.  The gain into these sections is controlled by the second section of the dual-gang "gain" pot; thus, this pot has a multiplying effect on gain when in 2-stage mode, which should give a wide range of control over the distortion tone.

The "pull raw" switch, borrowed from other Fender modification websites, effectively eliminates the tone-stack, producing something closer to the so-called "Tweed" tone.  Gain is higher without the insertion loss of the passive tone-stack; and since the tone-stack fundamentally causes a "mid scoop" no matter what its settings are, the "pull raw" switch leads to more prominent midrange.

"type M" gain section

This circuit is the same as the Marshall 2203 from the "low gain" input onward.  Thus, when the input section "voice" control is set appropriately, the overall signal chain is similar to the 2203 from the "high gain" input.

I use the available 4th triode section for a cathode-follower at the output, to buffer the passive tone-stack and level pot.

One of the circuit elements most responsible for the "Marshall sound", is the first triode section: this triode is set up as a "cold clipper", through having a relatively high cathode resistance (10k), and no bypass capacitor.  This section therefore clips the waveform asymmetrically, which accentuates the even harmonics, and this leads to a tone with a pleasingly bright and intense distortion edge, while still preserving readable dynamics and some of the original resonance of the guitar, similar to a "clean" tone (because half of the peaks are clipped, but the other half preserve their original shape).

In Soldano amps, I've read that a similar cold clipper circuit is employed, but the cathode resistor is even higher, 39k.  In this circuit, I provide a "pull colder" switch which brings the cathode resistor fairly close to the Soldano value: 37k.

Notice, in this schematic and in the one for the "F+" section, I mark a dividing border between circuitry closely based on the named original (there may still be slight differences), and elements I have added which are not part of the original ("BWK-M").

gain sections

Here are my notebook sketches, showing more or less what I want to do with the gain sections.  This doesn't show some of the surrounding bits such as the bypass relays.  My amp will have two "type F+", in positions 1 and 3, with a single "type M" in between (and the bypass switches allow any combination to be enabled, from none to all three in series, but always in the same fixed ordering).

I've drawn dashed lines to separate the portions of the circuits which are essentially direct copies of their namesakes (based upon public-domain schematics), from the supporting portions designed by me ("BWK-M").  (Take care to distinguish these dashed lines, from the ones representing conjoined dual controls such as pots and switches: I should have made them more different.)

The component values in the BWK-M portions, in many cases, are just my initial guesstimates.  These values "should work" to at least produce sound, but no doubt much adjustment will be needed once there is a working prototype.

reverb section -- initial component values

reverb section

Latest schematic for the stereo reverb/effects section:

As you can see, component values have not been fixed yet; this just shows the topology, with the new four-tube circuit.  Given the necessary 7th triode section for inverting the dry signal, I've used the "extra" 8th triode as a cathode-follower to more-effectively drive the reverb transformer.

important note re reverb and effect-loop phase

I haven't started to draw the "official" schematic for the reverb section yet, but I observe that in my previous writings here, as well as in many of the diagrams in my notebook, there is a serious problem: the "dry" signal is out-of-phase with the "wet", at the mix knob.  I had been planning to take the "dry" signal right from the input of the whole circuit section, i.e., non-inverted.  But at the position in the circuit of the "mix" knobs, the "wet" signal is inverted, having passed through one inverting gain stage.  After the mix pots, the signal is inverted again, so the overall reverb/effects path is non-inverting.  The fix is that the "dry" signal must be taken after the first inverting stage.

With my new conception that effect sends and returns need to be line-level and not higher, the effect returns will now share the same input circuits as the reverb tanks, so this phasing problem and its solution apply equally to the external stereo effects loop, as to the internal spring reverbs.

(Note that the "dry" signal for the mix pots, is not the same as either the "bypass" signal which appears at the outputs when the relays are in bypass mode, or the effect-send signal.  Both of these latter are non-inverting signals, as discussed in an earlier post on the reverb section.  Thus, unlike in the case of the effects loops associated with the active EQs, the send signal here will remain always-active.  Also, this effects loop is in-phase.)

Just wanted to get this important correction out there (mainly to myself, so I don't forget).  Schematic coming soon...

----

Ummm... more about this... dry signal taken from existing first stage will not work, because this stage is after the source-switch: the dry signal must not be affected by the source-switch, it must always come from the eq2 output position, i.e., the "natural" input signal for this stage.  It is the contrast between the distorted dry signal, and a reverb signal which is fed from an earlier, cleaner signal location, which the source switch is intended to enable.  If both dry and wet are fed from the earlier point chosen by the source switch, then there's no point to the later stages, they would be feeding nothing.

So... I don't like this, but the initial conclusion would seem to be that *yet another* 12AX7 is needed for the reverb section: a total of four.  Yikes.  I feel a great desire to get this tube count back down again, I was already unhappy with three.

One way to reduce the tube count, would be not to buffer the outputs of the mix pots.  This circuit only has one target (load) to worry about, i.e., the inputs of the power amps.  It's either driving the power amps, or it's bypassed.  Other circuit sections have to be able to handle a number of possible output loads, depending on which bypass relays are energized, and also depending on the position of the reverb source selector (which puts some extra load on whichever output it is monitoring).

My original idea for the power amps had the master-volume pots right at the inputs, then followed by tube stages.  But this will only work well if all the other preceding circuit sections have good, low-impedance tube output drive: if the output of the reverb section is "passive", coming from the mix pots, there will be poor performance directly driving the volume pots, and there may be a noticeable change (mainly, in sound level), switching the reverb between "in" and "out".  The power amps could perhaps be reconfigured to have a tube stage ahead of the volume pots; but if this requires actually adding a tube, then nothing is saved: it only makes sense if the reconfiguration can be done within the current tube-count (I'll look into it).

However, maybe the best way is for me to just, reluctantly, get used to the idea of four tubes for the reverb section alone.  The final stages in the reverb circuit, are closed-loop inverting stages, which give better performance for the resistive mixing networks, because the resulting virtual grounds at the tube inputs, ensure significant separation between the dry and wet signals.  Without the virtual ground, the resistive networks might have enough "leakage" that 100% dry or 100% wet settings are not attainable.

Well, that's my current stream-of-consciousness about all this!  As you can see, it is a work in progress, and significant issues remain in play, yet to be solved to my true satisfaction...

active EQ

Here's the latest schematic for the 3-band active EQ.

As you can see, I have decided to move the effects send/return positions to "inside" the boundary of the tube circuitry.  I.e., the send comes after the first tube stage, and the return comes before the last two stages.

Originally, I had planned to locate the send/return outside of basically the entire circuit, i.e., "right next to" the bypass relay contacts.  This would have meant that the send was always-active, even when the relay was in bypass mode: generally, a preferable situation.  However, due to the higher signal levels of tube circuits, jacks placed in these positions would not have been directly compatible with normal line-level audio gear; an external interface unit would have been required, just like the "Dumbleator" unit which is required to use external effects with Dumble guitar amps.  It's easy to pad down the send signal, but the return signal needs active gain to bring it back up to "tube" levels, and then the issue of inverted phase comes into play.

The clean solution, without adding more active circuitry, is to give up the always-active send, and put the jacks as shown in the diagram above.  The send can be padded as desired, including with an adjustment pot as shown; the return has a fixed, open-loop gain when the plug is inserted (the NFB path to the Bax tone control is broken by the switching jack).  The send signal is thus not always-active; it is only active when the section is selected "in" and the LED is on.  When the section is bypassed, the send is muted, because the input of the EQ circuit is grounded.

Note that there is one fairly significant disadvantage to my current effects loop arrangement: the external effect sees an out-of-phase signal.  As long as the external unit is non-inverting, its output signal is then inverted at the return stage, and everything ends up in-phase.  There should be no audible difference.  However, for certain applications (such as using external test equipment like oscilloscopes), it might be inconvenient or confusing that the loop signal is inverted.

There are two identical copies of this EQ section within my amp, named "eq1" and "eq2": one before the distortion, one after.  This distortion-bracketed-by-double-EQ configuration is a crucial part of my sonic concept; along with the nature of the EQ itself, i.e., active-Bax to give the capability to boost as well as to cut, and shiftable mid frequency to permit production of precisely-controlled formants.

input section

Here are the latest developments concerning the input section.  Also, this begins my first attempt to draw a schematic diagram for the whole thing (piece by piece).  I'm drawing by hand (obviously) in black ink on blue-grid graph paper, then scanning to PDF and processing the images with a combination of "NETPBM" open-source tools, and programs I wrote myself (originally as part of my pierced-foil art project, a different story entirely...).  Final result is a reasonably clear, and small, black & white PNG file.

A few things to note here.  I based the input jack setup on Marshall/Fender, i.e., one jack high-gain, the other with a resistor pad, but I made the pad resistors asymmetrical so that the low-gain input (2) pads down more than the original 1/2 division.  The high-gain input (1) still sees about the same 1M Ohm impedance, with about the same grid-blocking resistor (36k instead of 34k, with the values shown).  This front-end circuit seems to be an accepted "magic" set of values, used in many/most of the big-name (as well as the boutique) tube amps.  So I guess we'll stick with it -- other than making input (2) more useful by giving it the deeper padding.

In other images and text before today, I've been referring to the 6-position switch in the input section, as "HPF" (highpass filter), but now it's called "voice".  That's because it controls more stuff.  Originally, I based this control (HPF) on the Orange/Matamp "FAC" control, which selects the bypass capacitor on the output of the first triode stage.  Now, as you can see, I have changed to a dual-gang 6-pos switch, and the switch also changes the cathode circuit of the first triode.  There are three choices of cathode circuit, each with two choices of bypass capacitor: the "regular" value, and a lower capacitance (formed by two caps in series), which will give a higher cutoff frequency for the low-end.  The three cathode circuits correspond to Marshall 1987 low-gain input (positions 1-2), Fender input (3-4), and Marshall 1987/2203 high-gain input (5-6).  These are arranged in order of higher cutoff, i.e., decreasing low-end, as the knob is turned clockwise.  Normally, positions towards CCW would be used for "clean" tones, and positions towards CW would be used for "dirty", but of course the reason for the separate control is so that one can choose otherwise!  (E.g., one might wish to use a more-CCW position for a highly-distorted sound, if one were then going to dial-in a significantly non-flat response in the following active EQ module (eq1); the input stage would pass a full-range signal, which the EQ would then shape as desired.)

The second triode stage is an inverting stage with negative feedback; I've shown gain of -1, but it could be set a little higher if the loss through the passive components upstream is too high.  Most of the time, we'll want to keep the input gain set for clean headroom, and then use the switchable gain stages to produce overdrive; but we want to have enough gain to produce some reasonable overdrive with just the input knob, when desired (i.e., for the closest thing to "Tweed" sound that this amp can produce).

Hopefully, the inverting-NFB stage as above, will work well as the output stage for several of these modules.  Each module output may have to drive any of the switchable modules as input, or the inputs of the power amp if no intervening modules are enabled.  Getting all of the module gains and drive levels set right, so that the system works as intended in each of the possible configurations of module in/out (6 relays == 64 combinations total), will probably take quite a bit of experimentation, measurement, and component-swapping.

more design work on the reverb system

Here's the latest iteration of the front panel layout:

(That big blank panel on the bottom row, is for the "McGrath" logo.  I think this will be done with masking and black paint, so that the background will be black and the letters will be bare aluminum showing through.  Of course, for users of the amp, or for those who copy my design and layouts, this logo area also represents six or seven knobs-worth of extra panel space, for added circuitry.)

You can see the new four-knob layout for the reverb section (source, tone, mix L & R).

I've been planning out the circuitry to go behind those knobs, in more detail.  Looks like the system will need three 12AX7 tubes, not two as I originally hoped.  Two triode sections (one tube) will be needed for the reverb driver, in order to support the active tone/contour control.  The driver circuit will drive a single transformer, connected to both reverb tank inputs; I believe I will provide four RCA jacks from the transformer output, two in parallel and two in series, so that more experimentation with different reverb tanks is facilitated.  The reverb returns (L & R) will be amplified by non-feedback inverting gain stages (two triodes), and these stages drive the "wet" sections of the dual-gang pan ("mix") pots; the "dry" sections are fed directly from the original input signal (i.e., the output of the eq2 bypass relay), with no buffering.  The wipers of the pot sections are summed together through resistors, into the virtual ground of negative-feedback triode stages (two more, total six -- i.e., three tubes).

Because of the six-position source selector switch, the bypass wiring gets a little complicated.  Instead of one DPDT bypass relay like most of the subsystems, the reverb system needs two DPDT relays.  Both poles of one relay are used for switching the two outputs, L and R.  One pole of the second relay switches the input of the reverb driver, between the output of the source selector switch ("in"), or ground ("bypass").  The effect-send jack also feeds from the selector switch, but is not affected by the bypass relay: i.e., it is always-active.  And the other pole of the second relay disconnects the "dry" signal from the mix pots when bypassed, to reduce loading on this non-buffered signal.  When bypassed, instead of feeding these mix pots, the dry signal will be feeding the final master-volume pot sections: just about the same kind of load.  (The source of this signal might be any one of the prior subsystem blocks, depending on which ones are bypassed or in-circuit.)

To summarize:
reverb driver input: subject to source-switch; subject to bypass.
effect send jack: subject to source-switch; not subject to bypass.
dry signal: not subject to source-switch (always from eq2); subject to bypass.

Thus, the effect send jack can be treated as a "monitor" output, which can sample the signal at various points in the chain.  In addition to the obvious uses, this jack could be used to connect a tuner, a VU meter, other test equipment such as oscilloscopes or spectrum analyzers, etc.; or it could be used as a buffered line-out to drive other amps or signal chains, with the source switch providing the option for this signal to be clean or distorted.

The L & R effect returns are simply circuit-interrupting jacks, inserted at the last point before the output bypass relays (i.e., effect returns are of course subject to bypass).  Thus, there is no buffering or padding, of either the effect send or the returns; the external circuitry must be able to handle and produce tube-compatible signal levels (nominally line-level, but with possible very high peak levels).  I thought about putting the returns earlier in the circuit, so that they could be affected by the "mix" pots (and also they'd be tube-buffered); however, this would have made these inputs inverting.  This would have also required me to separate functionality and add another pair of jacks: power amp in L & R.  Instead, I can retain the dual functionality of the effect return jacks: they are also the power amp inputs when the preamp is off.  (By unplugging the reverb tank outputs, one can get access to high-sensitivity inputs which are subject to the mix controls, although probably only RCA jacks will be available.)

Depending on how much extra complication I want to add, there might also be a "post-tone" effect send jack, which sends the same signal that drives the reverb transformer.  Thus, the active tone/contour circuit could be used as a pre-EQ for external effects, if desired (and also, possibly my real motivation, it will be easier to monitor and instrument this tone circuit).  This post-tone send jack will be subject to bypass, like the spring reverbs themselves; unlike some guitar amps, the reverbs will not be driven when the reverb subsystem is bypassed.

So... it's obvious how the overall functionality of this subsystem works, right?  One can have either spring reverb, *or* external stereo effects, but not both.  The bypass relay (i.e., footswitch or front-panel switch) controls whichever is active, spring reverb or external effects.  The spring reverbs are operative until the external effect outputs are plugged into the effect return jacks: then the external effects take priority over the spring reverbs.  If desired, one spring reverb (L or R) can stay active, by plugging into only one of the two return jacks.  The effect send stays active, regardless whether spring reverb or external effects are being heard at the output.  The source selector switch controls the input to the reverbs and also the effect send, but not the dry signal.  Thus, the dry signal can have more distortion or a different EQ than the "wet" reverb/effects signal.  The "mix" pots vary between 100% dry and 100% wet, independently for each speaker, so the spring reverb can be effectively "panned" all the way to one speaker if desired, or placed anywhere else in the stereo mix.  Unfortunately, as discussed, the external effects will not be subject to the wet/dry mix knobs, for pragmatic reasons; stereo panning and wet/dry mixing will have to be accomplished by the external effects unit, itself.

reverb tone control

More thoughts on the reverb subsystem.  I think it will need some kind of tone control.  A bright reverb will sound more responsive and immediately gratifying, but many real acoustic spaces have a high frequency rolloff characteristic; thus, a darker reverb, although it may not call as much attention to itself, may ultimately prove to be more realistic-sounding and more supportive to a good guitar tone.

So at the least, a control which can reduce the treble frequencies seems important.  However, since the reverb already requires tubes for the drive and returns, the possibility appears to be available, to make the tone control active: i.e., able to boost treble as well as cut, with "flat" reliably centered in the middle.  This amounts to half of a Baxandall tone circuit, just the treble knob, with bass effectively hard-wired to "flat".

But, as long as we are adding facilities to flexibly alter the tone of the reverb in useful ways, within the space of one knob, we can make more options available by using a dual-gang pot with a pull-switch.  When pushed in, the control acts as a simple Baxandall treble knob, as described above.  When pulled out, the two gangs of the pot become Baxandall treble and bass controls, both moving in parallel: thus, when rotated clockwise, the control boosts both the bass and the treble, creating a response curve akin to the "loudness" curve on a home stereo, i.e., a curve which is mildly mid-scooped.  Rotating the control the other way, counter-clockwise past the 12:00 "flat" position, the bass and treble are cut while the midrange stays the same, resulting effectively in a mid-boost curve.  This pull-switch functionality is called "pull contour".  I have not decided whether to label the overall control "treble", or "tone".

So, the reverb module now contains four knobs: the six-position source selector switch, the pull-contour tone control discussed here, and the two wet/dry mix knobs for left and right.

taking shape...

 

overall shape & size

I'll be referring to the front panel diagram in the previous post.  Panels are 1+1/2" wide (I'm pretty sure: there's also the possibility to use 1+3/4" "U" channel; I'll only go to that if space requirements turn out to demand it).  Panels are separated by 3/4"-wide strips of wood (Poplar or Tulipwood 1x2s).  The longer horizontal panels (which are the three gain stage modules -- but other things can go here) are 7+1/4" long.  From this, the other dimensions are implied: the overall perimeter of the collection of modules (not counting the surrounding 3/4" edge framing) comes to 8+1/4" high by 18+1/2" wide.  There will be a 3/4" outer frame, and then the case of the amp may add another 3/4" to each edge, depending on my exact case design, yet to be determined in detail.

Thus, approximate overall width of the amp will be around 22".  If the amp is also about the same height, i.e., a square front face, with the module panels in the upper half of the face, this leaves about the same area beneath the panels, where a side-by-side pair of 8" speakers would nicely fit.

As for depth, the plan is to use the narrowest standard lumber width which will work: possibly 5+1/2", but more likely 7" or even 10".

The plan is for the speaker cabinet to be closed-back, for the tonal and power-handling characteristics that this imparts (but both stereo speakers will be in a single enclosure, with no baffle or partition between them).  Power-handling is a concern in particular, because I plan to use speakers that are just barely rated to handle the power that the amps will generate: this, to obtain speaker breakup and distortion at high volume levels, when desired.

However, into the closed speaker enclosure, other items will also need to fit, to keep the overall "square" profile.  The two spring reverb tanks will be mounted in the bottom of the case.  And, unlike most amps, in this design the tubes are "topside" of the chassis, but the transformers will hang underneath; i.e., the transformers mount to the underside of the partition between the tube circuitry, and the speaker enclosure; the wires pass through holes drilled in the partition (plywood).  Note that there are four major transformers: the left and right stereo output transformers, and the two power transformers, for the preamp and for the power amps.  (As noted elsewhere, the two power transformers are an extravagence mandated by the large number of preamp tubes -- and thus, the large amount of heater current required, by this overall circuit.)

So hopefully, the transformers will manage to stay cool enough, even though enclosed in an unventilated space.  If this is not the case, i.e., if cooling becomes an issue for some or all of the transformers, then some may have to be mounted "topside", with corresponding re-arrangement of all the other components; probably, the result would be a larger total size for the combo amplifier.

For the separate head/speaker configuration, I assume that the general layout will try to stay the same as the combo wherever possible, so this would point to a rather tall-looking "head", with the 20"x10" (approx) control panel, above a blank rectangular area at least 4-5" high, enclosing the transformers (and reverbs).  One nice thing is, the cabinets will be rather tall-looking, which may not be the most graceful proportion, but weight-wise, the center of gravity will be low.

design of active tube-based EQ

I've been working out the circuitry of the 3-band v-mid EQ, of which there will be two copies in this amp (the pre and post EQs).  I have been helped by several Internet "gurus" of tube-amp design, who I will credit in an upcoming post: there are a few excellent websites out there with tons of useful information for anyone working with tube audio.  Also, many of the famous-brand guitar amps such as Fender, Marshall, Orange, Vox, etc., have complete documentation including schematics, publically available (either from the manufacturers or from contributed reverse-engineering efforts by enthusiasts).  All of this has given me the confidence to design this amp, assembling the design from some directly borrowed circuits, and some created or heavily adapted by me.  In the latter category is the active EQ.

The issue is that the typical guitar amp treble-middle-bass "tone stacks", as found in the "F+" and "M" gain modules in this amp, are not able to really produce a midrange boost.  The passive tone stack circuit naturally produces a "mid-scooped" frequency response; twiddling the knobs changes the relative heights of the bass and treble peaks and the depth of the mid valley, but a basic scooped shape remains.  These response curves are an essential part of the "Fender sound" and the "Marshall sound", which is why I preserve the T-M-B tone stacks in the gain modules.  However, even amps with the scooped response in their electronics, will still ultimately tend to produce one or more peaks in the midrange.  These peaks, known as "formants", are crucial to the voicing of the guitar tone.  Various components in the signal chain, especially the speakers and cabinets, contribute to formant production, but in many cases these elements are hard to change; thus, a particular guitar and amp combination may have a distinct tonality, which persists despite any changes in control settings.  If the tonality is a good one, then this is not necessarily a bad thing; but some of us wish for more tonal flexibility, for a single system which can produce a great number of varied tone colours, perhaps including both the "familiar favourites" and also electric guitar tones that have rarely been produced or heard in the past.  To follow this latter path, for guitar, one quickly comes to desire variable midrange boost; and for this, active EQ is needed.

Studying the topology of various active EQ circuits on the Internet, both tube and opamp circuits, I've picked up quite a bit of useful information, though my own designs must be considered experimental and "naive" until I've done quite a bit more building and testing.  So beware...

One major advantage of active EQs, beyond the ability to more easily produce a mid boost, is that the level controls naturally end up wanting to be linear pots, as opposed to audio taper, and the midpoint of the rotation ends up being the "flat" position.  The basic characteristic of all the active EQ circuits, whether tube or solid-state/opamp, is that the level controls for each band select or "mix" between the input signal, and a negative feedback signal which is an out-of-phase image of the output signal.  By contrast, the corresponding passive EQ circuits will have level pots which mix between the input signal, and ground.  By considering the "virtual ground" principle, it can be seen that in the active EQ circuits, the midpoint of the level pot is where the circuit is in perfect balance, with the signal from the negative feedback taking whatever form it needs to, to completely cancel the input signal as seen at the wiper of the pot.  If there were no RC frequency dependence, it would be a simple cancellation with the same amplitude of signal at opposite phase; but with the RC action in the loop, e.g. the output will contain more high frequency signal if the RC network removes some from the input signal.

So with my 3-band design, the three level controls (bass, mid, treble) will each have a center-flat position.  The fourth knob will be mid frequency; now I am considering about a 20:1 range, of 200Hz - 4kHz.  It will probably take some field testing to determine exactly what range is needed.  The frequency knob will also have a pull-switch, "pull wide", which will widen the mid peak by changing one of the capacitors.  I.e., in effect, a choice of two "Q" values, though both are pretty low.

I believe I have worked out a good way to implement this EQ using two 12AX7 tubes.  There is nothing exactly like this on the Internet, but I've found enough confirmation for the various parts, that I think it will all work together as I am intending.  The design starts from what's known as the "active Baxandall" circuit.  This gives the bass and treble.  Then, I found an elabouration which provided 3 fixed bands, bass-mid-treble.  My contribution is to remove the capacitors from the midrange portion of this circuit, retaining the mid level pot and the fixed resistors at each end; thus, the overall resistance of the Baxandall circuit stays the same, which I hope will keep the bass and treble operating as they are supposed to.  The mid level control becomes a plain pot, selecting between the in-phase and the out-of-phase sides of the Baxandall network.  The output of the pot is buffered by another 12AX7 section, running as a cathode follower; this buffered signal drives a variable-frequency RC bandpass filter (Wien bridge circuit).  The dual-ganged frequency pot changes the "R" value in two points of the circuit; the "pull wide" switch changes one of the two "C" values.

The output of the Wien bridge BPF passes through a summing resistor, and recombines with the output of the Baxandall circuit; this summing point is the grid of the output amplifier, which operates as an inverting amplifier, and its output becomes the drive for the out-of-phase side of the Baxandall circuit.  Thus, as with opamp circuits, the summing point becomes a virtual ground, so the summing of the signals becomes "mathematically pure", and interaction between the signals is suppressed.

As I originally "borrowed" it, the active Baxandall and active fixed-freq 3-band EQ circuits were inverting, because they used a cathode follower at the input to drive the Bax circuit, and then the inverting-with-feedback stage on the output.  I have already added another stage, the mid-driver cathode follower above.  My additional changes are to add the available fourth stage as a cathode follower after the inverting stage at the output (i.e., a two-stage inverting amplifier, with better impedance characteristics).  And finally, the input stage changes from cathode follower (gain of roughly +1), to an inverting stage with local negative feedback, producing a gain of -1.  Thus, the overall circuit becomes non-inverting.

In my original design, I had planned to use a passive Baxandall circuit, and a buffered-but-passive midrange circuit, so overall there would be significant signal loss; thus, I planned to fit a gain-adjust trim pot at the input stage, to allow flat-gain to be trimmed to 1.  However, with the fully-active (feedback-based) circuit described above, I believe the flat-gain will inherently equal 1, so no trimming should be needed.

initial front panel layout

At an early stage, I started laying out the front panel of this amp, in order to decide which controls I want and how I want to lay them out.  The panel layouts are written by me, directly in Postscript: this is my preferred "graphics program" for things like this.  It is easy to change elements of the layout, such as the placement of controls or the choice of text for labels.  The original Postscript files (as well as PDF versions of the same) will be made freely available when this design is finished, as it is my intent to give away every aspect of this design as fully as possible.  I include a JPEG rendering of the layout here, to give a general sense of where the design stands, but don't depend on this image too much, many things are likely to change before anything actually gets built.

Each module is fashioned from 1+1/2" extruded aluminum "U" channel.  For convenience, I've made sure that no individual panel is longer than can fit on a regular sheet of 8+1/2"x11" paper.  Thus, DIY-ers can simply print the panel layouts and affix them to the extruded aluminum panels.  There's a whole procedure for doing this, in a way that results in a nice-looking and long-lasting panel; I'll detail the procedure somewhere down the road, when I start actually doing it for this project.  But the executive summary is: cut the paper to fit; spray a coat of white spraypaint on the panel; while the paint is still wet, carefully place the paper and let the paint adhere it to the panel (there's no chance for a re-do, so make zero mistakes if possible); press it down flat, taking care to work out any air bubbles; after a thorough drying period of at least 24 hours, cover with plenty of thin coats of clear-coat (clear spray enamel or polyurethane), allowing proper drying in between.  Probably the best time to cut the holes through the paper to match the holes which should already be drilled in the aluminum, would be after the white paint has dried but before the clear-coating begins.  If you're patient enough, this procedure can produce quite good, "near storebought quality", results (remember "NLQ" printers?).  (Thanks to the great Philip Williams for inventing this procedure.)

the modules

A list of the modules, as currently envisioned:

* input section: two input jacks, input level (pull bright), 6-pos HPF switch.
* active EQ: treble, mid freq, mid level, bass, LED. (2x)
* type M gain stage: gain (pull colder), bass, mid, treble, level, LED.
* type F+ gain stage: gain (pull 2-stage), bass (pull alt voice), mid (pull raw), treble (pull bright), level, LED.  (2x)
* reverb: 6-pos source switch, L mix, R mix, LED.
* bypass switches: a panel of 6 switches which replicate the footswitch functions.
* master section: NFB in/out, resonance, presence, master volume, mute switch, preamp switch, power amp switch, two power LEDs.

In addition, I have designed two other "horizontal" modules, which can go into the same slots as the gain modules; however, for my own amp, I won't use these:
* 5-band graphic EQ: LRC resonant circuits with tube opamp, center frequencies based on the Mesa Boogie Mark IV.
* dual semi-parametric EQ: similar to the variable mid controls from the four-knob active EQs, possibly with wider frequency range.

reverb and effects

This amp will have two spring reverb tanks, to produce a true stereo image.  Even with identical tanks, there will be subtle differences between the left and right signals, leading to a stereo effect.  But it will be possible for users to install different tanks for left and right, e.g., tanks with different numbers or lengths of springs, or different decay times, to obtain a more dramatic stereo effect.  The left and right return signals can each be separately mixed from full-dry to full-wet, enabling unusual stereo settings such as full reverb from one speaker and full dry tone from the other.

The spring reverbs are bypassed if external effects are plugged into the return jacks; the reverb and effects loop share the same bypass relay, and it's not possible to have both spring reverb and external stereo effects active at the same time (although one could plug a mono effect into one channel, while keeping the spring reverb in the other channel, for some interesting sonic variations).

For an unusual capability that no other amp provides (that I know of), the reverb and effects loop will have a six-position switch to select the "send" source.  In the rightmost position, the send will come from the end of the signal chain as normally expected, i.e., from the output of the second EQ.  But turning the control to the left will select progressively earlier points in the chain: the output of each gain stage, the first EQ, or the output of the first input stage itself.  Thus, depending on the settings of the EQs and gain stages, and depending on the position of this source selector knob, the reverb or external effects can process a much cleaner early signal, with later distortion and EQ colouration only applying to the "dry" signal.  There are not a lot of examples of this type of sound in the recorded canon, but I suspect it may prove musically useful.

In addition to the spring reverbs and the main (final) stereo effects loop, there are two earlier points where mono effects can be inserted: each of the two active EQ modules provides an effects loop, switched by the same bypass relay as the EQ.  Plugging an effect into the return jack overrides the output of the EQ, but two send jacks are provided, pre- and post- EQ, so the external effect can either replace the EQ, or can cascade after it.

The motivation for providing these earlier mono effects loops in addition to the main stereo loop, is that just like in the case of EQs, there can be a radical difference in the tonality produced by an effect when placed in front of the distortion stages, as opposed to afterwards.  Of course one could plug the effects into the chain ahead of the amp entirely, but then the guitar sees the load of the effect circuit, whatever that may be; with the arrangement here, even the earliest effect position still has tube buffering, so the guitar always sees the same input circuit (which is a one megohm load to ground, with 34 kohms series into a tube grid: i.e., the traditional Fender-style guitar input).

None of the effects loops has its own dedicated buffering; instead, I simply exploit good buffered "send" locations where they already exist in the chain.  Thus, the external effects units themselves are responsible for not loading their inputs too much, and for driving their outputs the "right amount" to re-insert into the signal chain.  And probably the most inconvenient part, the effects units must be able to handle "raw" tube-circuitry signals, which are nominally supposed to be at line level, but which can potentially climb much higher, into the realm of several dozens of volts peak-to-peak.  Tubes don't especially mind this kind of input overload, but solid state circuits certainly will!  (Effect sends from tube circuits often include some type of pad or divider, to reduce the signal level, and then of course added gain on the return inputs to compensate.)  Probably it would be more accurate to call these "patch points", rather than "effects loops", but I stick with the more widely understood terminology.

modular layout, gain stages, footswitches

In terms of physical layout, this amp will be modular.  Each significant function block will be built as a separate module.  Each module comprises a rectangular region of the front panel, containing the associated knobs and/or switches.  The module also contains a circuit board (not necessarily a PCB, however; wiring will probably be done point-to-point, using so-called turret boards); and on the back panel are the associated tubes.  Most preamp modules have one or two 12AX7 tubes; some may have up to four.  The modules are housed in segments of extruded aluminum "U"-channel, which forms the front and back panels as well as the "floor".  Thus, quite a bit of EMI shielding is provided to the electronics, even though I plan to build the outer case of wood.  The modular design is to facilitate incremental development, construction, and testing, as well as to encourage others to develop their own modules.  Each module will come with "flying leads" (wires), which are soldered to connection points on the amp chassis; this choice favours reliable connections over the convenience of rapidly swapping modules in the field.

The heart of this (or any) guitar amp are the gain stages.  This is where the bulk of any desired distortion is produced; even with so-called "clean" guitar tones, the mild distortion produced by tube gain stages is a critical element of the sound.  A Fender amp can produce a nice "clean" tone for a guitar; plug that same guitar directly into a hi-fi stereo amp or mixing board, and you'll hear a very thin and disappointing tone, lacking in the interest and "sparkle" of the Fender amp tone.  An electric guitar is really only part of a musical instrument; the formants and non-linearities imposed upon the signal by the amp or other electronics, become crucial components of the complete musical instrument whose sound we call "electric guitar".

In my amp, rather than having a single high-gain section to produce the distortion, I have elected to provide several lower-gain sections, which can be cascaded to produce just about any desired level of distortion.  Each section has gain, tone, and level controls, allowing subtle control over the character of the distortion as it is built up.  Alternatively, although this is technically a single-channel amp, different gain sections can be set up for different tones and can be switched in and out by the footswitches, affording some of the flexibility of a multi-channel amp, while also providing the ability to run some or all of the sections in series, simultaneously, for extreme levels of gain.

Initially, I plan to build two types of gain stage: the "Type M", which copies a Marshall 2203 preamp, and the "Type F-Plus", which is a Fender AB763 preamp with an additional switchable gain stage (i.e., either one tube with two triode sections, or two tubes with four sections).  The amp will have space for three gain modules, which I'll probably populate as F+ -> M -> F+.

The footswitch board will have six buttons, each a regular push-on/push-off metal footswitch, with an associated status LED.  The LEDs will be replicated on the amp front panel, with each LED colour-coded and located with its associated preamp module.  Each footswitch activates one of the six bypass relays: two EQs, three gain stages, and the final stereo effects loop.  For convenience in cabling, the footswitch unit will connect to the amp using a standard 8-conductor "CAT-5" cable, of any desired length (six switched lines plus power and ground).

the preamp -- pre and post EQ

The preamp of my new guitar amp will have lots of knobs and lots of tubes.  However, through bypass relays, it will be possible to cut the signal path down to a one-tube preamp with not much more than a level control, akin to early amplifiers like the Fender "tweed" models.

One key factor in shaping overdriven guitar tones, to me, is the capability to shape the spectrum of the tone (i.e., EQ) both before and after the distortion-producing gain stage(s).  The spectrum of the tone going into the gain stage has a big effect on the character or "texture" of the distortion; changes such as turning up the bass will not necessarily produce more bass at the output (indeed, the result could well be more treble and high harmonics), but the nature of the distortion and the playing-feel will be greatly altered.  Changing the EQ after the gain stage, on the other hand, will have the expected results on the final sound: turning up the bass will result in more bass being heard, etc..

Different guitar amps have historically placed their tone controls at different points in the signal path (e.g.: Fender: close to the input; Marshall: close to the output), and this is one big reason that different amps sound different from each other.  Many guitarists, perhaps unwittingly in some cases, have discovered the importance of being able to EQ the tone in at least two places in the chain; but since most amps don't directly address this, they use EQ pedals or other frequency-selective boost pedals, ahead of either pedal- or amp- produced distortion, and then they use the amp tone controls to shape the final sound.  Many of the popular distortion pedals distinguish themselves through the tonality they impart with built-in pre-distortion EQ.  For maximum flexibility, my amp will provide two full sets of active EQ controls (bypassable), before and after the gain stages.

The usual tone-stacks used in Fender and Marshall amps provide bass, mid, and treble controls, but due to the design of the passive R-C circuits, it is not actually possible to generate a mid-frequency "hump", just more or less relative proportion of "mid scoop".  A mid hump, aka formant, is a fundamental aspect of the "voice" of most musical instruments, including guitar.  Most guitar-plus-amp systems exhibit a formant, often chiefly due to the frequency response anomalies of the speaker cabinet.  But the only way to change such a formant is to swap cabinets.  Twiddling knobs on most amps won't make much difference to the formant.  Again, some guitarists employ frequency-selective pedals to produce or alter the formant shape of their systems.  EQ pedals are an obvious approach, but also the common trick of using a wah pedal kept at a fixed setting, has been used by many guitarists over the years to produce a dramatic formant in their tone.

Because of the importance of formants, I am electing to use a 4-knob active EQ, in both the "pre" and "post" positions of the signal chain.  This EQ provides fixed bass and treble controls, based on the "Baxandall" circuit, which enables both boost and cut of the fixed-frequency "shelves".  For midrange, a third boost/cut level knob is provided, along with a frequency knob (initially, I'm looking at a 10:1 range, from 200 Hz to 2 kHz, though this may change with experimentation).  There is no control to change the "Q"; i.e., this is a "semi-parametric" mid.

the outlines of a new guitar amp

So, on to the basic parameters of what I'm planning to build.  My primary concern is the preamp, because that is where most of the tone-shaping takes place, at least in the way I use guitar amps.  So what I'm designing will be amenable to a preamp-only realization (e.g., a rack mounted unit), or to the typical guitar arrangement of preamp and power amps in a box (the "head"), with separate cabinets for the speaker(s).  But for my own use, I want a small and portable unit, with limited power -- but unlimited flexibility.  So, my remarkably-complicated preamp will be wedded to a very small and simple power amp and speaker setup, in a single case: a "combo" amp.

My ideal amp will be small in size and power, but it will be stereo.  I find that stereo is crucial for reverb and certain delay effects.  Unlike the wide stereo image which is the final result of a mix in recording or live sound, stereo as applied to a single instrument such as guitar, serves a quite different purpose; as such, it is not usually important or even desirable to have a wide distance between the stereo speakers.  The stereo effect creates a doubling or chorus-like ambiance, similar to the double unison strings on harpsichords or the upper strings of a 12-string guitar. The two side-by-side speakers in a "twin combo" guitar amp are ideally suited for guitar-stereo; it's strange to me that there are not many examples of this (if any?) available on the market. (*) But then, that's why we are here!  (A number of twin-combo amps would be good candidates to adapt to stereo: especially those with four output tubes, which could be divided into two push-pull pairs, by adding a second output transformer and a few other components.  But here, we are designing from scratch.)

I am considering several different combinations of power amps and speakers.  For sonic reasons as well as size, I'm interested in speakers smaller than the traditional 12".  Probably for the combo amp, I'll try a pair of 8" speakers, similar to what's in the Fender Champ -- but with a closed box.  For the separate-head version, a twin-10" stereo-wired cabinet might be the thing.

For the power amps, I am considering three main possibilities.  If I end up manufacturing and selling these, I will likely offer all three options, and perhaps others.  To wit:
* 2 x 6V6 stereo (single-ended amps, a la Fender Champ).
* 4 x 6V6 stereo (push-pull, a la Fender Princeton and others).
* 4 x EL84 stereo.
In all cases, the power tubes will be cathode-biased, with selectable negative feedback; tube rectifiers will be used, for their voltage "sag" and slow power-up characteristics.  (If it wasn't completely obvious, the aim here is anything but "high fidelity".)

Given the number of preamp tubes I am planning to use (13 in the current design), it is hard to find a single power transformer which can supply all the required 6.3vac heater current for preamp and power amps, while also supplying the right high-voltage (many transformers I've seen which have enough heater current, are seemingly made for non-audio applications, and have inconveniently-high HV values, often 500vac or higher); and to employ tube rectifiers, the transformer must also provide one or more separate heater windings for the rectifiers, at 5vac or 6.3vac, since these must "float" at the high DC voltage level of the rectifier output.  One obvious solution would be to include a second, beefy filament transformer, to feed all the hungry preamp heaters.  However, given the need for two transformers anyway, I've decided to split up the power supplies more completely, so that there can be a separate power switch for the preamp, and one for the two power amps.  (The main complication here is that now two (or more) rectifier tubes are needed.  In fact, certain configurations might require three: one of the power amp designs I'm looking at (from the Eico HF-81 stereo amp) uses a pair of EZ81s for its quartet of EL84 output tubes, instead of a more regular choice like a single GZ34.)

The system will be set up so that either the preamp or the power amps can be operated independently.  This has certain subtle implications.  E.g., the low-voltage 5vdc used for the bypass relays, LEDs, and the footswitch, is controlled by the preamp power switch.  When only the power amps are turned on, there's no 5vdc.  The issue is that the left and right inputs to the power amps are the effects-return jacks, when the preamp is operating; hence, the inputs are only enabled when the stereo effects loop is switched in by its corresponding bypass relay.  In order that the effects loop is "in" rather than byassed when preamp power is off, this bypass relay is wired "backwards" relative to the others: it is by default switched "in" when off.  Sending power to the relay puts it into the "bypass" state.  (Even though the relay is "in" by default, the LED will only light up if the preamp is turned on.)

Instead of the usual "standby" switch, I plan to fit a "mute" switch, which simply grounds the power amp inputs.  (Chatter on the Internet has convinced me that standby switches lack benefits and are possibly harmful to tube lifespans.  A mute switch addresses the usual use of a standby switch, with fewer downsides.  One industry example I'm aware of: Matamp.)

I have found that negative feedback in the power amp section makes a significant difference in the tone, but there is not a single "best" configuration; different amounts of NFB sound good for different playing styles, speaker cabinets, etc.; so, a switch will be provided to turn off NFB entirely, and "presence" and "resonance" controls will be provided, to shape the spectrum of the NFB when it is turned on.

So, to summarize the controls of the power section, or what I'm calling the "master" section:
* preamp on/off.
* power amp on/off.
* mute.
* master volume.
* NFB on/off.
* resonance (LF boost).
* presence (HF boost).
All of these controls are active when the power amp is on, independent of the preamp.  Other controls which I'll speak about in upcoming posts, are only active when the preamp is on.

(*) Note: a quick check of the Internet suffices to show how wrong I was: there are actually quite a few stereo, all-tube, guitar amplifiers available today.  The first three I happened across were Magnatone, Orange, and Blackstar: all three are making nice, low-wattage, tube stereo amps for guitar, with the right provisions for stereo effects, etc..  So, it should be easy these days to "audition" the sound of stereo effects as applied to guitar, through closely-spaced speakers, if you're not sure whether you want to undertake the extra effort and cost of building a stereo amp.  Indeed, one of these existing amps may fully satisfy you, and if you've got the bucks, then I guess you're all set.
But nobody else has my preamp!  Hee, hee.

designing my ideal guitar amp

If you take a glance across my blog topics on this site, it may become clear that I am building a small collection of unusual musical instruments and equipment.  This will all be used to realize my musical vision, through live performance, and particularly through recordings.  Guitar has always been my "main" instrument, and electric guitar will figure prominently in the new sonic mix I am seeking to create.  Up until recently, however, my designing and building efforts have been aimed elsewhere, particularly towards finishing my Pandalon.  Consequently, I've been making do with less-than-ideal guitar sounds, produced with the help of DSP gadgets and whatever old tube amps I have laying around.  It has always been my intention to remedy this situation, and at last the time seems to be right.

In the past, I have obtained good results by assembling guitar systems out of various rebuilt and re-purposed vintage tube gear.  My most recent system (as of about 2000!) involved a homebrewed 2 x 12AX7 preamp, feeding DSP digital reverb/effects, into a 4 x EL84 hi-fi amp (Eico HF-32) and 12" closed-box speaker.  Tone was not entirely unlike the Vox AC-30, although cleaner from the power amp given negative feedback and the closed speaker -- and dirtier overall thanks to my preamp, because I love distorted and overdriven tones.

As I started to sketch out the outlines of a new guitar system, I soon realized that I want to design and build from scratch this time.  Given that decision, I realized that whatever I come up with, may be of interest to other builder-musicians.  Hence, this blog!  Enjoy...

Please note, some of the circuits I will present are my own design entirely (I'm proud to say), but others are either copied verbatim, or derived or adapted, from well-known circuits available publically on the Internet, from companies such as Fender, Marshall, and others.  In many cases, these circuits were probably originally patented.  The patents may well have expired by now; and in any case, I consider reverse-engineering or copying circuits for personal use, to be "fair use", and the same would apply to any reader of this blog who wishes to replicate my work for themselves.  However: if anyone (possibly including me in the future) wants to build amps based on this work, to sell to the public, then I simply warn that a closer look should be taken at possible legal problems with any of the copied circuits.  Caveat Venditor!