Tiger amp

[Doug Coulter]

Digging around trying to find some already drawn schematics for the tube audio thread, I ran into some words I'd written about the old Dan Meyer Tiger .01, with my plans for improvement, which I did wind up doing. I have a pcb layout for the new one, but am looking for the schematic file among my older computers (I did do this decades ago...). In my opinion, this is as good as you can do in an audio amp, or as good as has ever made any difference, in both "golden ear" and plain old measuring tests -- I've always been a bit agnostic on that flame war. We used ears and gear both, always. One informs the other -- the gear tells you want to listen for, and the ear tells you what to measure, I just don't see the conflict except as a way to sell magazines. To do the latter golden ear tests, we did something a little novel, which was to hang two very high quality mics (actually we tried several different types) outdoors, and listened to nature and an acoustic band inside through them. We could not find (at the time) any recording media that didn't swamp the errors in even the original Tiger design, much less the new one. In tests, we actually couldn't hear anything wrong at all that wasn't attributable to the microphones themselves.

Here are the words I wrote lo, these many years back just before I embarked on a year or so of tweaking and testing (which was after a long time, years, thinking about it). The basic Tiger design was wondrous enough, and intimidating to attempt to "improve" so it took awhile to actually find ways to make it better. I feel like the end result was close to the pinnicle of my analog design career, which improved all the numbers about 5x or more, and took the thing to stable down to DC and up to 4 mhz at full power.

I note VMike said he had oscillation problems with his Tigers. We did too. Turns out that the ground isolation washer for the input jacks is crucial, lead dress has to be as they described in the kit manual, AND -- you either need long, separate cables to the preamp, or put a couple of large ferrites on them (each one separately). You can't hard ground *anything* together in the original design, it seems.

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An Analysis of The Tiger 207 Circut





This document will attempt to analyze the Tiger circuit developed by

Dan Meyer in a tutorial fashion. Much baloney is spouted by "audiophiles"

about amplifier design, and this tutorial is designed to cut out the

unscientific, vague stuff and give a common ground for constructive

discussion. We will explore the circuit for this excellent amplifier

from input to output and bring up points as they occur. A good text

to read before proceeding is the Op Amp Tutorial in National Semi's

Linear Applications Book, which gives the math and theory behind some of

the statements made here.



Most amplifiers need some kind of "conditioning" circuit at the input.

The Tiger is no exception, and has one worthy of note. The input circuit

provides an impedance that goes down with increasing frequency, and never

becomes a high - Q reactance to either end. This helps insure stability in

the driving source, and helps eliminate extraneous high frequencies.

Many amp designers forego such an input lowpass filter so they can show a

pretty square-wave plot. However, most of us don't listen to square waves.

The distortion of most designs increases with frequency, as the loop gain

goes down. In the days when everyone's best source material was vinyl,

a shocking amount of above-20kc energy was often supplied to the power amp,

particularly when mistracking occurred. Why care? Distortion of even a

very good amp may reach 10% or more at 30 kc. While you can't hear that,

any distortion implies inter-modulation, and you CAN hear the resulting

low-frequency products. I believe that this effect is largely responsible

for audible differences in amps that test about the same. It is no doubt

a factor in the effect we all know about where an amp will sound better

with a particular preamp, and a preamp that sounds rotten with one amp

sounds great with another. Note that all the points marked "in gnd" on

the schematic are tied closely on the pc board, and tied to the input jack.

This insures that the negative feedback is isolated from high-current

ground-loop paths and is actually comparing the output with the input. A wire

connects the input jack ground back to the center tap of the power supply,

which is the "star ground" point. More on this later.



Proceeding to the active part of the input circuit, we see a rather

novel configuration. Rather than the usual single differential pair we

have two. There are several benefits. First, the input base currents

will tend to cancel, making it easier to provide good DC performance at

a reasonably high input impedance. Second, this configuration allows for

push-pull drive to be generated, rather than the usual single-ended drive.

This will tend to cancel out all of the even-order harmonic distortion

products right from the start. I'm a strong believer in eliminating

errors at their source, rather than using negative feedback as a cure-all.

" Get it as good as possible first, THEN use NFB to improve it!"

Why are there emitter resistors in the diff pairs? These provide

transconductance reduction in the input stage. A look at the National

Semi article mentioned above produces the surprising conclusion that a

lower input transconductance will allow a higher ultimate slew rate in

a closed loop amplifier. This is because a higher transconductance requires

a lower pole frequency to roll off the gain before phase errors accumulate

and make the loop unstable. The idea is to make the maximum amount of

current available to slew the pole capacitor without producing too much

gain. The LM318 uses this technique and was the fastest slewing op amp

of it's time. The downside of this approach is that resistor mismatch

will cause a dc offset. In audiophile land, we can afford precision

resistors, however. The main reason you want a high slew rate in an

amplifier is to avoid ever slewing. Slewing implies that the input stage

is in saturation (clipping) which will cause distortion. I do not believe

that you need a ridiculously high slew rate for good performance, however.

It's much better to eliminate spurious high frequency inputs, rather

than have response into the RF region. Who needs the possibility of tweeter-

frying high frequency oscillation, which is easier to produce from stray

capacitance at high bandwidths?!





The negative feedback circut is novel in another way. Note that

the large electrolytic cap (ugh) is NOT in the high current path. This

will tend to eliminate any distortion produced by the cap's "ESR". On that

note, why do audiopholes call that part of the impedance which isn't pure

capacitive reactance "ESR"? If resistance was all it was, no one would

worry about it much. Non-linear dielectric absorption in mylar and ceramics,

"diode effects" in electrolytics, etc are the real culprits in "capacitor

sound" and if there is no or negligable signal voltage drop across the cap,

you can successfully use even electrolytics in a good design. The NFB

uses a 47k resistor for the DC portion to match the design input impedance

in an attempt to produce a low DC offset. In the AC portion, we see the usual

voltage divider with a capacitor across it to roll off un-needed high frequency

gain. This amp is actually stable a a closed loop gain of one, which is not

necessary, but is nice, as high power signals have a way of coupling to

input signals, causing oscillation. The less gain, the less chance of this

occurring.





Note the 2.2k resistors in series with the current sources for the

diff pairs. This helps eliminate the "tail pole" caused by capacitance

in the current source transistors. This is also why the bases of these

transistors do not have a series resistor. Adding series resistance at this

point produces a miller integrator with the base collector capacitance, and

one wonders why you see a series resistor so often in lesser designs.





The voltage gain stage after the diff pairs are nothing special, except

that there are two in a push-pull configuration. When the front end is

well balanced these provide a nice constant current across the bias network.

The emitter resistors are larger than usual for this stage, to make input

stage bias current less critical, and to allow the use of small-signal

transistors for higher quality. We might also note that the less variable

the power dissipation in this and the input stages, the less the VBE of

the transistors will shift as the waveform goes up and down. You might not

think of this as an important effect, but it is the reason early opamps

latched up a lot, and can produce effects up to a few hundred hertz. Ever

noticed the "bathtub" shaped distortion curves in many amps? This is the

reason that many amplifiers have a distortion rise at low frequencies.

Now that some fairly sexy transistors are available in TO-220 packages,

we might improve our amplifiers by using them in the "low-level" stages.

This would also make it easier to insure that the transistors in a

differential stage are at the same temperature as each other!





The bias network is a fairly conventional (now!) VBE multiplier.

Dan Meyer intended this to be a kit, and maybe that's why there is an extra

diode on the pc board in parallel with the heatsink temp-sensor diode.

Wires break, diodes that are thermally cycled sometimes open up, and

kit builders sometimes put the parts in wrong. A failure in the bias

net will instantly cook almost any amp, no matter what the protection

circut, so he used an extra-reliable one.



The output stage is one of the places where this design really shines.

Firstly, it is configured to have gain. This immediately makes the job

of the pre-driver stages much easier, as they don't have to swing to the

power supply rails. This reduces the heating effects mentioned earlier as

well. The collectors-together configuration acknowledges the fact that

transistors are most linear in current mode. An emitter follower has that

logarithmic base-emitter voltage to current ratio IN SERIES with your

signal. Talk about a dumb idea! The gain of this stage is set by the

emitter resistor networks in the first driver transistors. Their ground

is referenced to the output connector for the most accurate "measurement"

of what the speaker load sees, and also to keep high currents out of the

input circut. The speaker connector has a seperate wire back to the supply

to implement this. The output stage has a gain of about 3, set by these

resistors. Note the 220 ohm resistors in series with the collectors of the

drivers. This protects the base-emitter junction of the next transistor in

line from excessive current. A darlington stage follows the first driver,

so it can use relatively high impedances, and a smaller, faster transistor.

The darlington's first stage contributes it's output current directly

to the output so it gets there without the delay of a slow output stage.

The 47 ohm series resistors protect the transistor if the prot circuitry acts.

This darlington allows the use of a 100 ohm base turnoff resistor for

the output device to help it turn off fast. Remember that this amp was

designed a long time ago and the transistors of today were not available

then. The MJ802 and MJ4502 used in this design had a then-amazing 2 mhz

Ft, which is still pretty good, but you could still burn this amp out

(just about the only way you could) with a high-frequency square wave.

This is because transistors have a storage time and don't turn off as fast

as you can turn them on. This could result in both output transistors

being on at the same time, a disaster. Looking at the place where the

output transistor collectors come together, we find an unusual pair of

resistors. These serve to isolate the stage from the output somewhat, and

provide some feedback for the biasing system. The .1uf caps across them

prevent a high frequency, high Q circuit from developing with the inductance

of the wirewound resistor. The isolation helps make the amp stable with

a short circuit as some signal can still get back to the first driver.

The principle of making each stage as linear as possible and using a small

amount of NFB in each stage before closing the loop is one I endorse

heavily.



The protection circuit is a conventional volt-amp sensor which shorts

out the drive when limits are exceeded. It's about the simplest thing that

works ok, but I hate them all. If they work, they inevitably don't allow

the full safe area of the output devices to be utilized, and of course,

they produce distortion when activated. My philosophy is to use such

large output devices that the line fuse will blow-- the power supply can't

melt them! With the better current-bandwidth devices now available, this

is possible without having to wimp out the power supply.



The output network has the usual resistor-loaded inductor and RC network

to keep the output stage from seeing extremes of impedance caused by your

( unknown ) load. It works ok, and that's all you need. Since I'm careful,

I sometimes omit this from my systems.





THOUGHTS ON POTENTIAL IMPROVEMENTS





Why try to improve on something that nearly everyone agrees is the

best anyway. ( Hear a tiger before you write that nasty letter! )

We audiophiles have always liked to push the limits, and though amps are

already the best thing in any chain, better is better. The Tiger had

somewhat limited power, and tended to overheat when playing loud into

4 ohms. I have a friend who likes to make 1 ohm ribbon speakers, and

that would never do as a load for a Tiger. Some of us like inefficient

speakers (not me!), and the 60 watts or so from a Tiger just isn't enough.

The old ampzilla (which was worse at low levels) was thought superior

at high levels, but I and several others are convinced that that was an

artifact of the higher power available. ( The AIR distorts at high levels

anyway, see Beraneks ACOUSTICS for Info). I just like to see the heads

turn when I put on a recording of a quarter being dropped, about the only

REAL source material available in these decadent studio-driven days!

Incidently, check out some of the movies available in VHS Hi Fi, which isn't

all that hifi, but at least many of the sounds are real, because the guys

who make them don't know a "studioed" way to make a car crash sound or what-

ever. They just crash the cars and record it. Novel Idea, Huh?

Anyway, I digress. To improve the Tiger circuit I suggest the following:



1. Use better, TO-220 transistors in the front end, and burn more current

to get a faster potential slew rate. These transistors should be bolted

together on a slab of aluminum to ensure temp tracking and a very slow

thermal time constant.



2. Match all the resistors! Dan got better than .01% distortion with 10%

matching......



3. See above for transistors. These should be matched for different

parameters depending on where in the circuit they are. VBE and current

gain ( at the operating current ) are the important param's for the

diff pairs. Hfe should match npn-npn and pnp-pnp, as well as

npn-pnp for the front end to really cancel base current at the input.

The pre-drivers and drivers should match if possible too. These

transistors should also be as fast as possible (really throughout).

Dan used transistors with about a 60 mhz ft, and much better are now

available.



4. For more power, parallel more outputs. The Crown DC-300 does this

successfully with separate emitter resistors for the output transistors,

and it should work here too. Then maybe you could get rid of the protection

circuitry.



5. After the above, you could DC couple the thing throughout, and probably

make the main pole frequency higher for even higher slew rate. I've

thought a lot about making a DIGITAL offset adjuster that would sense

output offset during quiet periods and send a correction signal to a D/A

converter to adjust offset. This would give good DC performance even

in the presence of pre-amp offset and eliminate the nasty caps! I feel

that we can do a good enough job without this, however.



6. (And most controversial!) Change the output devices to FETs. Yes,

it's true that there are no complementary FETs, which is also basically

true of transistors. However, you could match transconductance in a

number of ways, unequal "emitter" resistors, for instance, and FETS

are FAR more bulletproof than transistors. FETs don't have a second

breakdown effect, period. They are also MUCH faster. I can turn a

40 amp fet on and off with < 30 ns risetimes, which is only possible with

delicate RF transistors that cost a LOT more. Remember, it is the

falling loop gain (necessitated by slow outputs) that causes distortion

to rise at high frequencies in most all amps. Here we have a chance

to make everything fast, which means we don't have to roll off the

gain as soon to prevent phase errors from making our amp into an

oscillator. Fets also parallel much better than transistors from a

practical point of view. Incidentally, has anyone seen a .2 ohm pot

anywhere? I thought not, but you could make a tapped resistor out of

pc board track for the purpose of getting rid of even order harmonic

distortion in output stages! Anyway, you would have to make a few

other changes to use FETS in the outputs, such as in the bias

network. You might use led(s) in place of the diode, for instance,

to add some constant (bandgap) voltage to the diode's. You don't

want the bias to go down with temp on FETs the way you need to

do with transistors. Note that the bias feedback inherent in the

Tiger circuit makes this easier to get working. While it's true that

FETs aren't as linear as transistors, especially at the rails, most

people like the "tube" sound anyway. I think the "tube" sound

is partly due to tubes "graceful" logarhithmic clipping, which is

what FETs will give. Also, most tube amps seem to have a better

phase margin than most transistor amps, which means less ringing,

cleaner high end rolloff, etc. And of course tubes are FAST.

Don't try making a 30 mhz transmitter with the output device from

your Transistor amp, but you can with just about any tube! Or any

FET, if you can handle driving it's input capacitance.



I've made some changes in the basic schematic to reflect what you

might do to make a FET version. This must be considered incomplete, but

note the following points:



1. I no longer close the loop to gain of one at HF. ( internal loops too)

2. The bias net has one or more leds in place of diodes. ( I'll test this)

3. No more protection circut.

4. Output darlington now is emitter follower, as FETs won't swing the

rails otherwise. This transistor will need a better heatsink.

5. Some resistor values are different, and more will change during

the optimization process, to reflect the much lower transconductance

of the fets (1.6 mho vs approx 10 mho for the transistor version)


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Anyone have a copy of the original schematic? I probably do, somewhere, but it might take a month of looking (full time!) to find it at this point. I don't even have a clue which of 5 buildings it might be in, and I've never been super organized...

I didn't wind up using big die front end transistors. One reason Dan had that output stage with gain is that high quality low level transistors of the time, and still when I was doing this, tended not to be able to handle the higher voltages needed in a design where the drivers had to swing the rails. I replaced the silly series R's and zener across the front end with real voltage regulators, as the resistors had tended to burn out and create spectacular failures, for one thing, and I was never comfortable with those input rails kind of floating.
One of the big tricks in my improvement was matching of the outputs. At the time (and maybe still) there were no really matched N and P fets, with the N ones always having higher transconductance so....cheap trick...I used a small source resistor in the N side only to reduce the distortion without feedback. I also munged the bias circuit, (and never truly got it right) for the different characteristics of the fets. The result was something that tended to try and heat the outputs up to a fixed temperature, so the bias current was fairly high when at low
levels. When you were cranking the thing for long periods, the heatsinks didn't warm up much more than that, and the bias current went down a little, but still stayed good. I felt at the time that rather than fool with that forever, just go with it -- when one of these was cranked enough for that to matter, you were into air-distortion range (and possible ear damage) and you sure couldn't hear the effect of temporarily reduced bias -- it never went close to crossover distortion anyway. I'll add more here as I dig around and find it -- would be a good thing to capture what's in the older computers in my network (easy now with big new disks in the newer ones) so I can reclaim the space they take up anyway.
That's going to take some backing and forthing because of course the old disks are IDE and not the faster versions of that. Hopefully I have at least one running well enough to
read them and get on the network...

At one point we were running a software development shop here and the "Good for that time" computers tended to accumulate, I must have 20 or so -- now not so amazing, one of them is even a P-II. A couple are worth keeping, as we got the special P-III tualitin chips Intel custom made for the LA School system. They were the forerunner of the chips later used in laptops, and only draw 8-9 watts at 900 mhz or so. They had much faster level 2 cache (full speed) than the normal pentiums of the day, to boot, and it really made a difference.
For a person on solar power, that's pretty sweet if you can stand the fact that you have to use older software that doesn't have today's bloat and the assumption that there are cycles to burn.
But that's another rant -- the stuff we wrote wasn't bloated one bit and wasted no cycles, and so got incorporated into a lot of products out there (still!) because it gave better performance to the products that used it than their competitors could manage. Can you imagine writing in assembly language today? We did when it mattered, and it usually did for signal processing kinds of things, or getting a pentium to be able to eat continuous data flows from even an 8 mhz uP blowing out 115k baud...most PC's would not handle that at the time unless you kept bursts to below the uart fifo size with long times between bursts!