lab jc

there are so many definitions of simplicity in the analog electronic world that the description is close to meaningless.

on the engineering side, it is generally used to satisfy certain design preferences or bias. rarely is the method “simple”. there are also varying kinds of electrical simplicity in a physical sense. i like the one that involves a simple transfer characteristic with a first order low pass well outside of the audio band. that is not so simple to accomplish, as it turns out. another one i like is integrated functional simplicity, where many jobs are accomplished at the same time with the fewest tricks. such as, using the natural characteristics of materials and devices to accomplish more complex functions… it looks so simple when it’s done and working! before that, anything but simple. how it fits together is absolutely not simple, even though the end result might conform to several different kinds of simple. there are more kinds of electrical simplicity than this. and many other opinions about it, too.

on the advertising side, “simplicity” most often refers to a commercially constructed ideal of elemental “purity” and “innocence” (both equally meaningless in real terms, but dangerous and seductive fantasy words) and is therefore valuable. you know you want it, you dirty humans! this is often portrayed, at least in the states, in a kind of creepy love story between absolute evil and simple innocence. i love that story… my favorite story! blue velvet and little red riding hood. “mulholland drive”, is one of my favorite movies. also. “let the right one in”, in the original swedish.

on the consumer side, “simplicity” is most often connected to physical appearance. things made from unpainted metal that have one switch and no knobs, for example. with a remote made from the same material with the minimum of buttons. for some reason silver or white, reflective colors, seem to have have become markers for “simple” as well. a mirror is simple? your face is simple? you ARE simple? hmmm. just as gloss black is “elegant”, and polished gold is “rich”…  you occasionally see gear that doesn’t even have an indicator that shows it’s on, but those things don’t generally sell well outside of a particular fetish tribe. gadgets with this kind of appearance infer an inner simplicity, that rarely exists. it’s the idea that matters here, not the reality. remember the stereo in “a clockwork orange”?

for the diy builder, and many consumers, simplicity is mainly a mythical quality that is hidden away inside the chassis and the careful choice of certain types of parts. there is an intense sub genre of parts fetishism associated with it. it’s an opaque “black box” kind of quality. “pure”, but also somehow “classical” in a newtonian steampunk sort of sense, unsullied by 20th century relativism or quantum mechanics. and even perhaps more than a little touch of “right”, as in the sense of “correct”, but with an older ozone flavored alchemical feeling around it. a secret “right”, that only a few enlightened minds are privy to… sounds complicated to me!

i suppose i am not exempt from all this, except i have nothing simple to market or consume… i don’t have anything simple to sell! especially not myself. no, not simple. it isn’t simple for me to design or make anything… and i can honestly say i have not put any effort into making it simpler, either. and anyone accusing me of being simple would probably win my heart over, and the right to walk all over me. that would be music for these ears! oh, to be simple for someone! but, sadly (?), it has NEVER happened. i am beginning to suspect it won’t. and, i AM usually right (my wife doesn’t think so).

for the diyer, electronic simplicity has mainly boiled down to a reduction of components. low parts count = “simplicity”. hahhahahahahahahahahhaah! what a ridiculous concept. it isn’t true. that kind of “simple” means simple to assemble! gear with a low parts count is very rarely simple. each component does many jobs and usually not optimally. the character and nature of the materials plays a much bigger part in the performance. the transfer characteristics of each function are much more fixed, and there is less flexibility in operation, even if the range is relatively “wide”. you usually can’t so easily edit or adjust a low parts count design for optimum, and the nonlinear aspects are features whether you want them, or not. often these sorts of gadgets can’t even be hooked up to just any load. that does make your systemic choices simpler as you may be limited in terms of application. this is not to say that simplicity is a myth. just that it’s more complex than a low parts count implies.

think about what happens at the edge of something… anything. it might only be moving wind around a corner, a jewel bearing in a watch, or a piece of rock skipping on a pond. a waterfall. a meeting between two different states. this is a naturally turbulent interaction. a very low parts count for sure! yet, one of the most complex interactions known… something that still challenges physics, math and modeling on super computers today. something that is still studied and the knowledge base is expanding. it isn’t “complete”, in other words, whatever that means…

there is also the simplicity of the fewest possible changes of “state”. many years ago, there was a man who would go to washington square park in nyc on summer weekends with an edison cylinder player. a big one! he dragged it around with an over sized vintage radio flyer, himself looking kinda like a tall wimpy poopdeck pappy. he would play caruso, and valentino accompaniments, flapper stuff, all sorts of fun stuff… i somehow remember “i’ll never see maggie alone” (“there was her mother, her father, her sister and her brother…”). it sounded fantastic! much better than i had imagined and actually within the limits, authentic. behind the scratchy bandwidth limited old mechanical recording shtick, there was a palpable and substantial experience that electronic recording doesn’t have. i am not alone in this judgment. it is an experience shared by sid smith and dick sequerra too. we had this conversation over pizza many years ago. eddy reichert and i have also talked about it. why is that so?

great sounding player and horn speaker

dick’s straight up answer was simple: fewer changes of state between the original energy and the reproduction. less loss, plain and simple. no, the bass and treble were gone… and surface noise was added. but the middle was “closer” to the original by virtue of the simpler chain of transformations. kinetic to storage media, and back again for playback. electronic media requires many more transformations than that! each one losing subtle info that cannot be retrieved. mechanical recording has a sense of audio verite that is built upon it’s simplicity. if you haven’t listened to a victrola or edison player recently, you owe yourself the experience. that is a kind of simplicity. but it isn’t so simple to describe… it also isn’t optimal.

for another example: input, interstage and output transformers are complex components electrically. they add levels of complexity to a circuit in excess of the reduction in parts they allow. this is not to say there is no place for such components! no. just that they are absolutely NOT simple. depending on the quality and application of a transformer, you minimally have added a 4th order filter with large values of parasitic energy storage (some large enough to qualify the trans in question to have as many as 4 more poles). you add time to the process at hand too, and distribute the changes over the audio spectrum in a varying way. this one component could actually be replaced with dozens of other components and still do the same job in the same way. yes, i know. this argument really catches in the craw of many. but it can be demonstrated.

vacuum tubes, because of the simpler physics that underlie their operation, all by themselves, have a “simpler” transfer characteristic than say, bipolar junction transistors. tubes work because of the physics of heated conductors. semiconductor physics, heated or not, rely on much more complex interactions between materials that are themselves modified to express their useful characteristics within carefully engineered ranges. this statement includes a comparison based on the added complexity of the use of coated cathodes for tubes, instead of pure or thoriated tungsten, for example, with P or N doped silicon. coating a cathode with oxides (calcium, barium and strontium are the common ones) employs electrical chemistry that is related closely to the science of semiconductors. especially when things go wrong! but before i go too far down a road in the opposite direction, i want to make a point. there is a big difference between building a machine for a particular job with easy to apply components (op amps, like transformers, are way easier to use than tubes, and have simple predictable results…) than it is to really consider all the relationships between the parts and weave them together in a way that appears to have been “meant” to happen. it still gives me a great deal of satisfaction to consider that the musical instrument business has had a terrible time trying to replace the push pull 6V6 guitar amplifier. they have had 40 years to do it… it just refuses to die and go away. culture is important, after all. that is easy to say, but not at all simple.

i sign off here with an idea for a really simple circuit, that isn’t simple at all. ever wonder what a triode “sounds” like all by itself? here is a way to do it… yes, all those resistors are needed to get the thing to be stable, and biased. and, the op-amp servo, keeps the DC output well under a mV. you could get rid of the servo and DC couple it to a tube amp with a large cathode resistor on the first stage, just to allow for the wandering output… but nothing else. no headphones – no DC amplifiers.

this would make a good non-inverting line stage. or maybe a headphone amp…? it needs a bigger power tube for typical modern phones actually. perhaps a 6550 triode wired? but totally ok for beyer dynamic DT880S (600 ohm) or similar. you could rearrange this with your favorite or maybe some old tube you wonder about… some safety provisions would have to be added to protect the load from offsets or failures… more about this later. actually, i will build this. interesting…

some will complain about it being a cathode follower and that you won’t “really” know what it “sounds” like… fuck you, moron!

here is the gain stage version, although it inverts phase and will have some gain, so you still won’t know what it “really” sounds like… since whatever goes in, will still be transformed into something else afterwards. so fuckin there!

the bypasses and compensation are probably absolutely necessary and tube dependent. yes, it’s so simple, you will have to treat every tube you put into it as a unique situation and opportunity. not a one size fits all kinda gadget. but fun… totally fun!

i have been in south korea working on silbatone stuff and i wanted to post a little something about a cool circuit i’ve been playing around with for phono. i’m going to do a mic pre front end for ribbon mics with it soon. but it’s very very quiet (just under 1nV per root Hz) and sounds lovely. this is not for beginners or probably those who have a moral problem with hybrid circuits… but it is interesting anyway.

cascode circuits offer a way to build your own pentode out of two triodes, JFETS or BJTs, and have some advantages over standard pentodes… lower noise and no screen current. many have tried JFET/triode cascodes in an attempt to get even lower noise but still hang on to a “tube character”. i would have included myself into that group some time ago but now i believe this has it’s own sound and i prefer it for phono and mic pres… blasphemy! but i have still had some reservations about the harmonic spectrum of the typical cascode. tube or N type JFET (i love the 2SK146 and 147) with high Gm type triode (WE417 or 437, etc.) on top. it is ok for blues, jazz and rock and roll… but i haven’t been totally happy about the classical music thing, or the dense noise thing yet. cecil taylor records or cop shoot cop just doesn’t doesn’t sound it’s best… it’s just a touch too hifi, and that is a four letter word in my book! i think pentode operation has a better timbre even if the distortion is higher. Gm operation of a pentode or cascode can simplify the distortion spectrum over standard operation. i have mentioned this before. but cascode, even single ended, has a slightly forward exciting sound typical of circuits with a touch more odd than even harmonic content… even more so balanced. the distortion is very very low! but the timbre isn’t perfect.

but i have found my new toy! hybrid folded cascode.

as far as i know, folded cascode was done first in audio by john curl (?) with JFETS. he does folded complimentary balanced circuits that are a study in symmetry and beauty. they also work as good as it gets. i think he is still the master of it, but the technique has expanded into chip design for memory and many many other jobs too. a folded cascode is a complementary amplifier in cascode which requires an added resistor to leverage the I/V conversion between the two halves. because the two amplifiers can be independently biased and loaded, it is possible to “tune” the response with more finesse than with a more standard cascode. using a P type JFET folded into the cathode of a triode is a way to have a complimentary amplifier with a pentode like transfer charateristic. the gain can be modest or high, depending on the configuration, and it can also be arranged as a Gm amplifier. i know of no one who has made anything commercial with this other than us… but now you can make something with it! have fun!

here is the basic idea as i have been using it. this is a Gm amplifier (loaded with a constant current source) and a resistor to ground. in this case, i am using a very large valued resistor (1 meg) and getting 60dB of gain… (the sim says 68dB). that is mainly due to the JFETs… solid state varies 1000 times more than tubes in it’s characteristics.

and here are some pix of the circuit in the flesh… the other tube you see is a bendix 6900 used only as a buffer for the folded cascode. the empty tube socket eventually gets a WE437 for make up gain after the RIAA filter. that goes beyond the scope of this discussion. today we are only looking at the first stage, which is a hybrid folded cascode circuit using 4 each 2SJ74 PJFETs and one D3a wired in triode. what you see is the transfer characteristic at 10KHz with 3mV input. the scale is expanded for detail (uncalibrated).

badass. i know it’s not good to brag but this is really very good. actually, it performs very nicely, but most importantly, it sounds just lovely! pix to follow of the entire phono preamp…

this next installment of the EH 12AY7 mod series is going to deal with the power situation. the original has a very clever (if i say so myself) and very economical power supply completely organized around the idea of compact size and market/regulatory acceptability… (much of the pedal world functions far far away from the high voltage world of vacuum tubes). it makes all of the various voltages needed from one 12 volt AC wall wart, with few parts and without any switchmode anything (still too noisy for cheap analog). while fulfilling the design requirements, it is not ideal in terms of the best performance. mainly, because of the limits of the wall wart and supply. also, the method for distributing all of the various currents interconnects them… a drag on any one of them, really affects them all. this is mainly a problem because of the phantom supply. certain modern mics draw a fair amount of current from the phantom. some of them, more than is available for the preamp…

the upgraded circuit improves the isolation and the drive capability of the buffer, but also requires even more current than the original. the solution for all this is a more optimized and dedicated set of supplies for the various jobs at hand. this way we can optimize the various functions. let’s start with the big one: the high voltage B+.

double regulated 220 VDC supply

ok, here you see a pretty straightforward solid state regulator. it has three basic sub-circuits. it is based in it’s entirety on National Semi’s© venerable applications propaganda for the LM317. i am a big fan (although they didn’t write this stuff, bob pease and jim williams are such heroes for me. because of guys like them, the quality of the writing and depth is routinely excellent. the national applications notes are deeply useful and well done. the guy who did the work for the LM317 was very matter of fact and sometimes really funny). an added mosfet current limiter ahead of the chip, and a transconductance regulator after, make it much more sophisticated and safer than originally proposed. some small refinements to reduce the likelihood of high frequency oscillation, and better short circuit protection have also been added. a few words about all of this are warranted.

the high voltage rail calls for 220 Volts DC. this was provided for with a “pi filter” following the rectifiers in the original. in order to reduce hum and variability of the rail, we can arrange for a separate supply and then regulate it. mic pres often deal with tiny signals and the balanced differential circuit used in the design of the mic pre only has so much power supply rejection, and it works less well the higher in frequency you go. a tightly regulated supply takes care of that! however, an interesting problem does arise. high feedback regulators, such as the LM317, or even any of it’s improved versions (such as the wonderful LT1084, 85, 85…), also make high frequency noise that isn’t so easy to compensate for. all amplifiers add noise, but especially those with gain, so it isn’t that remarkable. it’s analog! deal with it. those chips have a LOT of gain. the unregulated supply did have some hum and sag, but no high frequency crap at all. now you know why the DIY world is so dependent on passive solutions… it takes work to deal with this crap! the remedy is not that complex. following the chip with a “no gain” regulator, the stiff regulation is mostly retained, but the noise is reduced by a huge amount. we can also decouple the final regulator with a small film cap, which will respond to transients with aplomb.

let’s have a closer look at the sub-circuits involved. the first mosfet is your basic source follower, with the reference voltage derived from the regulated output of the chip regulator downstream. the 100k resistor and the 15 volt zener make sure the gate is never more than 15 volts away from the output of the chip regulator. the 20 ohm resistor is sized so that if there is a short circuit at the output of the reg., the increased voltage drop across it will begin to cut off the mosfet and protect the LM317. this could be bigger in this case because the total current draw is so small, but since it will be used for stereo, and i am lazy…  the mosfet will heat up a lot under shorted conditions (a decent heatsink is needed on all the chips), but normally doesn’t get hot. the LM317 has a maximum voltage rating of only 35 volts. by combining it with the mosfet in this way, it can handle 100s of volts.

how is that? the LM317 and it’s family of adjustable regulator chips are “floating” regulators. the built in error amp works to match the drop across the reference resistor (in this case, R6), which is connected to the output, with it’s internal voltage reference. in this arrangement, a better more accurate reference is used instead (the LM329). the point is, this drop can be set without a direct ground connection, hence the “floating” moniker. the LM317 can source up to 1.5 amps provided it has a good heatsink and is protected from shorts. the Linear Technology© versions have better performance all around, but do cost more too. i use them all the time for low and high voltage regulation.

the final sub-circuit is very important. sometimes called a “capacitance multiplier” or a “transconductance regulator”, it is essentially a source follower (cathode follower for the vacuum tube folk) with a high impedance voltage reference on the gate that is bypassed heavily with a large electrolytic. this is important. the time constant needs to be LONG. that will multiply the effective “capacitance” of the output by the Gm of the device. a small film cap at the output can be made to behave as if it is orders of magnitude larger. handy. and it has no gain! if arranged in a stable configuration, the output noise can be in the nanovolt range. in this case, the uVolt range is good enough! we are talking about a 200 volt supply! 220V/1uV is 10^9 noise reduction! this is an improvement over the 2mV hum level of the original.

in the case above, i have deliberately injected 100mV of 60Hz hum on the 250 volt “unregulated” input. the 0.8 mV PP output ripple across a 10K load (22mA) speaks for itself (you will need to download LT spice to sim this). mosfets have the Gm and low rON necessary to do this right, but tubes can do a fair job of it too! as long as the load is constant, the supply won’t sag. additionally, while the impedance of a 6V6 or 6L6 cathode follower can’t match the mosfet, it would  be fine for line level stuff, or a screen reg in a push pull amp.

some will complain that it isn’t a tube regulator. i don’t care. tube regulators can sound great! but they are not capable of this level of performance. at least not without a large increase in complexity and cost. and you will never reach the really low noise level. however, if you are willing to go hybrid, there are some interesting possibilities… even just following a tube reg with a mosfet “no gain” reg will do quite a job. that’s for another time.

the heater supply on the original pre amp PC board is regulated and can be kept as is. the prudent upgrade is to feed it from it’s own transformer and filter. perhaps, just removing the chip and tiny heatsink from the board and mounting it on the chassis or larger heatsink will be enough improvement there… we’ll see.

the phantom supply is next.

okay, in the last installment, i outlined 5 areas of improvement to the EH 12AY7 mic pre, to explore and an initial design for how to deal with some of them. these were, gain, isolation, a lower output impedance, distortion and noise, and improved power supply in support of the other issues. i intend to get into it, but first, i am going to inflict some more wandering brain on you… it’s my blog! go make your own.

you still do hear a LOT of tongue wagging and brouhaha over the “subjectivist/objectivist” dualism. i want to say something about this, because it has a bearing on what is happening here. i hope i don’t overdo it, but it was bound to require some clarification sooner or later. it is a work in progress, but the basic flavor is bitter, with a touch of bicarbonate of soda.

“subjective” critics of the traditional engineering approach, to solving the issues of electronically reproduced sound, tend (an understatement) to dismiss wholesale the significance of measurements and analysis based on research and testing. it makes no difference that all electronic gadgets have always involved the use of “high” technology, often based on new science, for the present day, or even in looking back. audio engineering is described by this camp almost as if it were alchemy or magic. for them, it IS! and, they are apparently afraid to know more of how that magic works. it is so prevalent and the group is large enough that this perspective is heavily reused in marketing many highly engineered commercial products today. ironic isn’t the right word to discuss this. schizophrenic is a better one. the difficult thing about this for me is that at the center of this perspective is one very important misunderstood and misrepresented truth.

“objectivist” or “rationalist” critics of the subjective camp self righteously point out the delusional aspects of this. but often have very little self examination of their own. in fact, the wall of outrage directed so intently at the herd on the other side of the river, seems to allow them to avoid this altogether. they are, as a rule, as dense and unpleasant as the others are, and even more defensive because they feel science makes them right. “it’s in writing, mathematics, it’s been measured, for crying out loud!” the status of their situation is seemingly more complicated than the others, because some of them have an awareness of the truth at the center of the other side’s weird universe, that they lack, and yet don’t want at all for themselves. even though it’s true. the less sophisticated ones are just assholes.

so what is this truth? it’s the thing they have in common. that they are obsessively fixed on the meaning of reproduced sounds… and desperately intent upon control of that. but, no amount of objectifying will limit, reduce or expand the range of what is possible. music is not a signal. it isn’t a bit depth. it is a cultural practice. it is subject to all of the messy motives people have. if the gear was really and truly “neutral” (i have a great deal of difficulty with the choice of words here) and music passed on through all the changes of state between microphone and loudspeaker, without the subtlest change… really transporting one from the present location to a site far away in perhaps both distance and time. well, no, it’s not possible in an “objective” sense. that is a ridiculous idea. that is the “subjectivists” truth. although they are generally clueless as to what it means.

reproduced music is not “real”. and whatever “reality” it may have once had, is so socially constructed these days that it can best be compared to fashion photography or advertising. “objectivists” seem terrified or angry that even reproduced, music could be so wildly representative and that the true measure might actually be in emotional or anthropological terms and not scientific ones at all. we live in the post modern time and the “facts” are that fake is every bit as important, or more so, as “real”, for millions and millions of people. the music business is built on that idea. the argument is a case for the emperor of clothes, or ice cream. not math. and resentment over it. what a miserable point of view. the “subjectivists” are at least more fun, even if they are full of shit. when it comes to reproduced music, fun counts for more than misery.

chefs and cooks mitigate technology, science and art day in and out. no one gives them a hard time about it. it is clear that the preparation of food is both rooted in cultural and technological history and innovation. it is also clear that a good chef needs good eaters to be appreciated. if you don’t like shellfish, who cares how well it’s made. yes, marketing and commerce confuse things here as everywhere. but a great plate of fish is very difficult to argue over, if you like fish. this clarity is utterly lacking in the playback world. marketing runs that world… it’s almost all lies, based on their own fear and resentment. and it doesn’t matter that much if it’s “objective/subjective”… they’re both missing the point.

enough. microphone preamps, even cheap ones, have a job to do. to get the signal from the transducer (and it can be a range of very different transducers) and get it into a size and form that lends itself to recording or post production. but it also has an aesthetic: to be “musical”. ahh! how infuriating…

the one “aesthetic choice” i make here, and it is a somewhat controversial one for the tradition, is that this cheap mic pre runs open loop. there is NO error correction applied. what is, is what is, as they say. now, that is not to say that i do not care about distortion or bandwidth… measured performance is very important to me. however, there is more than one way to skin a banana.

the first two areas of improvement were gain and isolation. fortunately, there is a component we can draw upon that fills the bill. a step up transformer. trannies get a bad rap in all but two places, diy hifi and recording studios! and that is funny, because it couldn’t be for more opposite reasons: in the diy world, its retro sex appeal and love of the object in itself. for the recording engineer, it is simply practical. transformers afford isolation from the hostile common mode environment. not just the canceling of it, which is the world of the instrumentation amplifier, but real honest to goodness isolation. accidentally connect the wall mains to it? who cares? flourescent lights? big schmear. 80 feet of mic cable next to a gas generator and a motor bong… no problem! that kind of practical is hard to ignore.

i bought some 1:3 SUT that appear to be tamura units from yamaha mixers from the 70′s. removed from equipment and sold on ebay for a reasonable price. let’s look closer at what they can do…

here is pix of a test set up. the signal comes from my marconi generator to the red and black primary, the scope is and shield ground are connected to the blue and white secondary.

my test setup is a generator, oscope set up in X/Y with the generator on the primary and X ch, and the trans secondary on the Y ch. the AC voltmeter is on the input to make sure the generator is at the same level when i switch frequencies. sorry for the blurry pix. they were all taken with an iphone.

here you see the 25mV RMS input level, and a 200mV  plus change PP output. 1:3.

1KHz @ 25mV RMS input, here's what you get...

25mV RMS

what you want to see is these next images which show the transfer characteristic of the transformer at 100Hz and 20KHz. note the phase shift (oval shape) at 20KHz… quite reasonable actually. and finally a 2V RMS input gives 8V and change PP out without compressing (you would see curvature or bending over). there is both cartesian and X/Y examples. good to know it will survive a closely placed SM57 on a snare…

25mV RMS in @ 100Hz

25mV RMS in @20KHz

now, the important thing here is that the trans steps up from a 600 ohm load nicely and the X/Y is a nice straight line to the right… and no wiggling or funny ground loop stuff visible. this is a good trans, especially for 20$ for the pair. you can expect to pay more than this generally, but good deals do come up.

now cinemag and jensen both make good brand new versions of this, as does lundahl and sowter and others. for vintage stuff, you can use whatever you can find, like i did. these worked out just fine. tamura is good stuff. adding this to the input of our micpre will add both isolation and gain. and they are linear and have enough headroom for the mics they would likely meet. all that is needed now, is to remove the input blocking caps and wire this in instead. it will be necessary to revisit the bass cut feature later, but that can be added simply by using an appropriately sized film cap switched in or out between the trans and the H pad.

2V RMS in @ 1KHz

2V RMS in @ 1KHz

the next change has to do with noise. in order to minimize distortion, i increased the voltage across the FETs by biasing the 12AY7s up, until i got the lowest 3rd harmonic. in this configuration, that turned out to be 18 volts. i would have guessed 7 or 9 would have done it. but the combination of these FETs with this tube worked out that way. this is NOT the lowest noise arrangement.

above 3 – 3.5 volts of drain source voltage, FETs produce an increasing amount of flicker noise. very similar to johnson noise but for different reasons. at or below this drain/source voltage, there is very little. so i have lowered the bias to 1.8 volts. this should put 3 volts at the cathodes of the 12AY7s with 2.5 mA between them. 1.8 V happens to be the breakdown voltage of a yellow LED, which is also very quiet. some kind of LED will make for a good voltage ref for the bias. the distortion is a little higher, but it is still very low (0.1% for 8V RMS out) and the pre gains some S/N (another 5dB). the step trans helps here too as it adds very little noise of its own and 9dB of “gain” ahead of the active stages. the increased current through the cascode helps a little too, and we are on the way.

the lower output impedance of the buffer stage can be accomplished 2 simple ways: increase the current through the tubes; and change to a beefier output tube. both work well and are easy to accomplish. we can change the cathode resistors to 10K each from 33K for a threefold increase in current… and we can use a 12BH7 instead of a 12AU7. the heater circuit will need to be adjusted to accommodate the extra current required by the 12BH7. this will make for a considerable improvement in output impedance without any fancy tricks.

ok, so the first four areas of improvement have had some solution proposed, and the way forward seems plausible. the power supply will need to be flushed out a bit to support these changes, but we can also improve the noise performance by regulating the B+rail and by making a better 48V phantom supply. that will be next.

i think this will become a running post.

ok, i left off “part 1″ with a brief explanation of how and why an “affordable” (let’s just say it: cheap) commercial tube mic pre came to be. some notion of the design issues, along with some of the issues of amplifying balanced signals, staying balanced (in order to reject unwanted environmental noise and interference sources), and above all economy. an effort to design “good bones” into the gadget also leaves us a chance to explore the possibility of building upon this design as an affordable way to get somewhere substantially better. this idea is something that can quite often be expanded to many older professional products, as well as some unusual currently produced things.

from the previous segment, if we simplify the circuit so we can see the “bones”, it is fairly obvious that we have a differential cascode gain stage, followed by a pair of cathode follower buffers. a very simple 2 stage circuit. a constant current source in the common cathode forces a high degree of balance between the two halves. the gain of the circuit using the original parts is about 56dB in practice. this is an “average” and was arrived at using the audio precision “portable one” owned by EH. the spice version predicts 60dB. i take this as a home run and proof that the models are pretty good! the gain is largely determined by the transconductance of the FETs, which is modest. this is enough gain for most mics, but definitely not for ribbon mics, or some older condensors in any job other than close placement. i am not a fan of close placement, but it is a lucky thing that almost every other recording engineer is. still, gain is an area of improvement.

small offsets and AC common mode stuff between the inputs of the diff can be passed along this circuit if one “side” of the output gets loaded down more than the other. as mentioned previously, the 12AU7 buffer does not have a low enough impedance to be “immune” from loading effect. this is an open loop design, so there is no error correction at all. a lower output impedance would be an improvement. it is the case that some “prosumer” gear, namely recording interfaces, that have economical methods of gain adjustment at their inputs, do not have completely symmetrical input impedances because the gain is adjusted with negative feedback, and the “balanced” inputs are subtley different at certain gain settings. this is typical where the input circuit uses an inverting opamp stage although why is beyond the scope of this installment. suffice it to say that as the amount of feedback goes up (the gain goes down), one input’s Z goes down (the inverting) and the other (non) stays the same. if no effort has been taken to account for this in the design (a sign of economy), offsets and common mode can be coupled through by the gain of whatever it is in front of the recording interface (the mic pre). this economy is acceptable generally because the impedances commonly encountered are isolated from the outside world with instrumentation amp circuits and other methods to extend the CMRR of typical op amp based mic pres. but this simple tube pre has none of that! it’s an open loop, “let it rip” kinda thing. it works fine as long as the output sees a balanced load, but if it doesn’t, you can get hum and ground born noise coupled through from input to output, as well as from the power supply. so another area of improvement would have to be isolation.

and finally, noise is a pet peeve of mine… way more than harmonic distortion. nonlinear and enharmonic distortion is rarely nice in pre/amps either. and it is so often the case in cheap gear the S/N ratio and distortion spectrum are roughly considered. it is “cheap” after all. as long as these specifications fall into a certain range, no further effort is spent and the “sound” becomes fixed by this limit, plus any nonlinear/harmonic attributes part of the system. most of the op amps used in the business have a class B output stage and are also run LEAN at the input. there is so much feedback, it is assumed that all forms of nonlinear distortion are “taken care of” and the noise floor is determined by the Gm of the input transistors (the bias) and the various resistive elements (all impedance creates noise) of the design. this is fantasy. of course useful, but still not what many consider to be the case. many of the most commonly used op amps add crossover distortion particularly to small signals (the output stage has to turn on…), such as those encountered in a mic pre. and of course the noise floor can be impacted negatively with small signals (noise) that just cross the threshold of “on”.

another more controversial issue concerns the classic op amp, which has a gain – bandwidth relationship determined roughly by the miller “compensation” between the first 2 stages. this means that if the gain is 100dB at 10Hz, it is often only 32dB at 20KHz open loop (6dB per octave drop). the feedback error correction is not uniform for the bandwidth of the particular arrangement. there is much less feedback at higher frequencies always… also more distortion and noise. the general opinion is that this is unimportant, as far as the audio bandwidth is concerned. and for cheap stuff, doubly so. it is a good idea to look at the noise and interference sensitivity of the design very closely so as to reduce these things… a wideband open loop design can equal or improve upon this, and need not involve the complexity required for having even more gain and then losing it with error correction. do it right from the beginning and you don’t need to fix it afterwards.

you can see an open loop op amp (no criticism of it… it is typical) above. compare with the open loop sweep of the EH12AY7 mic pre below…

the gain is MUCH less in the tube gadget, but the bandwidth isn’t bad at all. the S/N for a 4558 or TL074 diff with the same gain and a 6.8K input impedance isn’t much different! it’s still 70dB roughly, for a 3mV RMS input signal.

ok, so the fourth area of improvement would be the noise and distortion spectrum, which ain’t bad already… this does imply one important last improvement: the power supply. these five areas are where i will focus my energy. below is a simplified starting point. i have removed the phantom supply, bass cut, phase reverse and monitior sidechain temporarily, just for visual simplicity. next installment will explain what has been changed and why.

to answer a question i was asked from the phillipines, i am posting this…

it is a direct coupled feedback scheme (important for me) and can be DC coupled further up the line using techniques outlined previously. yet another interesting thing about pentodes is that the plate voltage can go very very low and still be useful, providing the screen environment is optimal. i have gone as low as 10 vdc at the plate and 160 at the screen with certain tube types. i have built and  tested this circuit as the reverse of another attempt with a very low screen voltage. the trick of having a low plate voltage on the first stage and a more “normal” plate voltage on the second means they can share the same screen supply. a fabulous simplification. if the screen is low impedance, there is a beneficial dc feedback there. because of the cascaded pentodes, open loop gain is around 82dB. closed loop is 25dB. the performance is excellent out to 2 MHz. i am working on an all tube VCF (voltage to frequency converter) and need an “operational” amp with a particular stability. this gets very close. not there yet though.

it can be used for more mundane audio related things, or even as a regulator, with a change of buffer tube.

i can only measure down to -80dB from the 10 volt RMS range with the gear i have… i am making some things to help out. but 3rd harmonic is at least -80dB down with 80 volts peak to peak out (2 volts pp input).

i hope this shows that you can go the “other” way, too! (low plate voltage/high screen voltage)…

enjoy!

this is not part 2 of the mic pre lab.

as a response to a question on the joenet, i am posting a slightly simplified practical version of the silbatone linestage. if you know anything about tube lore, you will recognize alan blumlein’s classic current averaging gadget in the cathode circuit of the 5687/6900. sometimes known as the “garter” circuit (where did this come from?) due to it’s visual similarity to a pair of men’s suspenders. i imagine blumlein pacing in front of an oak desk in his. it still works well today, all these years later. in this arrangement, you see the 1930′s revisited in a thoroughly modern line driver. open loop, the distortion is 70dB down at the 2nd and drops from there across the spectrum. if your trannys are good, expect magnificent results. silbatone uses stuff you can’t get or probably can’t make. but you can use whatever you can get. i would say a pair of lundahl LL7901s and a pair of LL1660PP will fix you up in stereo at a very fair price. if you want to go DEE lux, give slagle, tribute, or your favorite tranny man (cross dresser?) a call.

the biggest difference between what silbatone makes and this schematic is that we currently use a silvercore cobalt amorphous cored stepped transformer attenuator and not the H pad you will see in the schematic. you could use one too, and not use the H pad i have drawn. kristof’s tranny sounds really really great, but, like any tranny, it has trouble with DC across it (when switching) and it needs to be loaded right. there are no tubes which will be perfectly balanced so this circuit will pose a problem for any variable switching thingy you put between the grids. a few words about this: inductors store energy. that is what they do. it takes time to charge and discharge. that is what we take advantage of in filter circuits… so thank goodness something does that job.

it isn’t always a good thing. i am all for inductive attenuators. but i am not at all for cascading inductances one after the other. the problem is lumped poles (multiple storage facilities). it is well understood that one transformer is minimally a 4th order bandpass filter. if you are using 2… you now have a 8th order filter, etc. if you have any appreciation for what that means for the time domain, you will understand why this idea catches in my craw. add another… whew. what we really wanted was to use a slagleformer at the input. yes. that’s what it is called. it’s a very special tapped autoformer attenuator. but, as you can see, we can’t here because the inputs float 20 volts above ground. isolation from the input is a must in this circuit. i will not use a 1:1 tranny followed by an autoformer in a circuit that also uses an output transformer to drive an amplifier that also has an output transformer. no, i am not sakuma. never will be either, and that’s ok. but MJ loves inductive attenuators. so we are changing the circuit in the next iteration so that we CAN use a slagleformer. it will be the only inductor at the input. but it won’t be this circuit anymore.

the finest attenuator i have ever seen was made by siemens. it was a broadcast plugin from the 1960′s. they are occasionally available in europe, but i have never seen one in the states. i don’t at the moment know what it is called. it is a transformer attenuator for mixing with many taps on the secondary. the trans was made by haufe. these taps are shorted with a carbon element which has a rotary slider that makes contact with both the resistor and the tap points. on the front face are markings for the various attenuation levels that correspond to the taps. the resistive shunt allows for going in between the points as well as preventing ANY possibility of ringing by shorting ALL the taps. you can return to fixed attenuation settings for remixes and archive work. the last element of it, which impresses the hell out of me, but is SO anti hifi: the output of the attenuator is buffered. with a transistor buffer. this is how to do it right, in my book, from beginning to end. i would use a tube buffer, myself!

update: peter sikking has identified the gadget in question as a telefunken W690D. he is also the one who told me about it first. it is superlative.

updated update: peter sikking is the one who described this to me, but alexander kliegl showed me a siemens W291. these seem similar? my excitement was completely due to peter’s influence because i had no idea german broadcast made anything like this.

in any case, the H pad is an excellent solution. this way, you can use a 1:1 transformer and have it constantly loaded at any attenuation level.

i hope this will prove interesting. the nickel choke in the B+ line is so important to the sound. the things is great without it, but it is fabulous with it.

which brings me to the circuit:

as an introduction to what goes into a mic pre, i thought i would start with something very accessible. no vintage collectibles. later on, i will go into some fancy stuff.

some years back, i designed a mic pre for electro harmonix. mike matthews wanted it to be based around one of the “other” reissue 12A_7 type tubes that were new production from reflektor (the tube factory he mostly owns in saratov). partly as a way to sell more other types of tubes. the 12AY7 was a natural choice. i love 12AY7s. the specifications for this pre, and this is where the rubber meets the road, were that it had to fit in a standard EH tube pedal box, it had to use a CE approved wall wart, and the total cost of the bill of materials precluded the possibility of signal transformers. i was sure we could do it and not use an opamp anywhere in the audio path. the retail price mike was shooting for just under 200$. i wanted it to be all tube, and use a high voltage supply. it wasn’t going to be easy.

now there are a lot of cheap ass “tube” mic pres in the “prosumer” market. for example, art© makes a really cheap one that has a tube in it. the tube does nothing good, but you can get some interesting grungy sounds out of the thing… for vocals, with a shure dynamic, it’s actually ok for punk, black metal and even hiphop. and it is really cheap. did i remember to mention that it doesn’t cost very much? don’t waste too much time trying to record something subtle with it though. it has no decent clean sound and is a fair bit on the nasty side. it’s an opamp based mic pre with a 12AX7 with the same 9 volt supply on the plate and on the heater (heater is normally 12 volts…) too. there’s no point expecting a lot. the tube is arranged as sort of a signal conditioning gadget that adds distortion and EQ. that is what the industry thinks tubes do. this sort of approach is used by korg, and others, as a way to get a tube “inside”, without a high voltage supply or spending a lot of money. the tube markets the product well beyond what it actually is worth. we are talking about the low end, mainly, but you’d be surprised how many of these sorts of products do well? well, maybe not. as configured, the 12AX7 has a gain of 0.3 – 0.7 (no, it is not a cathode follower), and not all 12AX7s work in the circuit because the operating point is on the edge of conductance. it’s like a diode voltage divider. rolls off the top and adds some child’s law flavored 2nd harmonic distortion. they sell a lot of them. in any case, that was the “competition” we were looking at. mike will probably never sell as many mic pres as most of the makers that focus on that, but they will never sell any fuzz boxes either. he will sell some mic pres. the total cost of the art© thing, before marketing and labor is probably 12 dollars or so… maybe 15$? it was machine assembled in china, and there isn’t much labor in it, but a great deal of marketing. at the time, i’m guessing the 12AX7 cost maybe 6$ wholesale in large quantities (no, you couldn’t buy them for that unless you ordered 10,000 + pieces). it was a tough act to follow. the chassis, pc board and parts consume whatever is left of the budget after they bought the tube. the tube was a third or fourth of the production cost. i hope you keep this all in mind as i continue?

okay. mic pres need certain things just to qualify as such. of course the basic house cleaning was figured out, as well as facilitating the use of different kinds of mics: the feature set took shape. the cost of the unit still stood as a sheer cliff between me and what i wanted to do. but, that is what it is to design consumer electronics generally. i prototyped a few different front ends and a big problem immediately raised it’s head. balance. this is a transformerless design and balanced inputs and outputs are part of what is required to interface with gear upstream and down. without a transformer to isolate the audio from the hostile electronic environment outside, a straight up simple tube diff front end, could NOT produce the common mode rejection necessary. even with a fet current source, the best i could do without adjustments was about 40- 50dB, depending on the tube. i do not design ANYTHING that requires a particular tube. matched tube. special tube… etc. it’s against my religion. it has to work with any tube. the other cheap pre’s on the market are not either very impressive in this regard, but i wanted to do it better. this was the first mic pre EH would put out. cheap or not, i wanted to make an impression. adding a balance pot added 20dB more rejection, for a while anyway. but wasn’t an acceptable solution. no user adjustments, that’s also in the rules of engagement (a prescription for misery if there ever was one). i had an auto balancing circuit i had designed using tubes in another project long ago, and i thought i could translate it to solid state for this… but the added complexity ate up board space i needed for conveniences, and budget. i had to do something simple. i cheated. the 2N3958 is a dual matched Jfet with a guaranteed CMRR of 100 dB. i made a hybrid cascode diff with fets on bottom, the 12AY7 on top. the jfets weren’t cheap, but i found a source in japan and it was cheaper than the balancing servo. a LOT cheaper than a transformer. it’s a tiny part too, which saved space. and i didn’t have to use an opamp, which would not have had any better CMRR in the price range we were dealing with, and would not have sounded the way i wanted. mike matthews and his people are really good at finding parts for less. that is key for success. i am sure, by now, they have a better source.

the next problem was the power supply. in order to sell things internationally, there is a heap of regulatory hurdles to jump through that is both mind numbing and nasty. nearly all of them are set to hinder or bleed manufacturers from outside a given trade zone. the usa does it too. in europe, the CE is mainly aimed at limiting the onslaught from china and india, and protecting local builders. but it affects US products as well. it has very stringent standards when it comes to radiated and generated emf at the wall. the simplest and most economical way to get around it is to use a CE certified wall wart. then, your gadget only has to comply with the rules set for input/output emf and ground safety. you also have to sacrifice one of every product you make to the testing procedure and cannot group them. this makes the sacrifice cheaper, if you use a common power supply for all of them. getting analog gadgets passed is rarely difficult, especially if you remove the power transformer (the whole purpose of the wall wart). you may hate wall warts but the industry loves them. the other standard that has to be dealt with now, is ROHS. which means it is heavy metal free. lead free solder, no cadmium or mercury anywhere. i am one of those that hates it and keeps a supply of 63/37 eutectic for my own personal use. but to do business, you have to use tin/antimony. it sucks.

mic pres do not make much radiated noise…  the linear supplies don’t either. however, the design needed 220VDC at 12mA (B+), 48VDC at 8mA (phantom), 12VDC at 350mA (heaters), 12VDC at 50mA, 6VDC at 20mA and 18VDC (bias). right there you see why art’s little pre just used 9V for everything, even the tube. there is no wall wart that can do all that. so, right away a decision was made to use a 12VAC 1A wall wart. the 12VAC could be directly rectified to make the various little voltages… but could also drive a 12V:220V transformer turned “backwards” to make B+. the 48V phantom supply was the last remaining problem. clearly it had to come from the B+. but here a little transistor regulator could be configured as a “pop free” supply: that ramps up or down from ground to 48V over a short period of time. none of the cheap pres out there had that. it is really annoying to switch it over and forget to mute things first. BANG! one additional issue is that electro harmonix uses many different types of DC wall warts. from 9V to 40V. most of them use the standard barrel connector boss© and others also use for pedals. the possibility of mixing the AC wall wart up with a DC one and blow up something was obvious. we (me and john pisani) also then made a decision to use a different connector: 2 pin european DIN. a most hated connector if there ever was one… but cheap! no one would be able to dumbly plug 12VAC at one amp into their fancy harmonizer or digital delay and cook it.

a word about the output stage. it is 12AU7 cathode followers off the cascode plates… and this is cap coupled using bypassed electrolytic caps! 47uF allows for “driving” 600 ohms. no, it does not have really enough cajones to do that with aplomb. but it sounds good. a bit dirty and old fashioned. i see this as a feature. if you intentionally load the output with 600 ohms, you lose gain and add distortion. really useful R&B sound. yes. heresy for the hifi world. totally ordinary for the pro audio one. these caps are low esr types made for switching supplies. a good tip for you hackers. these 2 caps were near the top in terms of costliness in the BOM. i think the pair of them were almost the same price as the toroidal trans. the bypasses are mylar 0.1uF.

here you see what i did:

a few comments are in order.

mic pres minimally need, in order to be useful, the following features: bass cut (6dB per oct. from between 50 and 80Hz, for vocals mainly); phase reverse switch; phantom supply (+48VDC for condenser mics); some kind of clipping indication; and a pad. there also needs to be some kind of adjustable sensitivity built in for the wildly varying signal level typical of switching from different types of mics and also from close placement. i chose to use an “H” pad with a pot in front of the gain. this pre is open loop. you can’t actively reduce the gain, so the pad is very important. after some development, it became obvious that an aux out would give flexibility for monitoring. we added this with an opamp using the signal at the output of the tube pre.

here are pix of the board…

for the top view, you can see there are 2 boards: a main board and a piggy back board. there is a ribbon connector that joins them. you can see the cut blue wires, which is where the 12VAC goes in. you can see the 10V:220V toroidal transformer used to step up the wall wart current to B+ level. the power supply components are gathered near the bottom of this view. there is a 12V regulator for the heaters and opamp supply. the audio is in the middle and the control stuff on the breakaway board. the quad opamp uses 2 sections for the aux out buffer, one for gain (X1.4), and the last as a clipping indicator driver.  you can see the 3 DPDT switches for the phantom, bass cut and phase reverse switches… as well as the green LED to indicate the phantom is on, the amber LED clipping indicater, and the blue “power” LED. the pots are the input “H” pad and the aux volume level.

on the underside you can see that this is a 2 sided board with ground planes on both. you can also see the neutrik PC mount connectors which we did spring for, and we still came in at cost!

there has been some good discussion of this pre over the years since it appeared. the tape op crew got ahold of it and many liked it… some did not. the ratio is pretty good, for to against. i think it still sells well. some problems have surfaced too, but all of them honestly related to the low cost. there are no low cost preamps that do it all. it could use a little more gain just for ribbon mics. one unforeseen issue: there are a number of popular low priced computer interfaces for recording (notably the delta 1010, for example), that do not have truly balanced inputs on particular settings. this means that hum can be coupled from input to output of the pre, in certain situations. it can be fixed using a transformer coupled direct box… but i had hoped to avoid this. i am really satisfied with the end result. i don’t think you can buy a better pre for the same money or even close to double.

ok, so why have i gone through all this? partly to demonstrate the thinking behind its design and just how much cost is a part of the calculation. but also, because it doesn’t take much work to modify this into a very much better mic pre… something that would hold it’s own in any studio. the most important components and features are all here on the board. you would be hard pressed to find all of this for less than the cost of a new one in the box. 2 of these modded, some i/o transformers, and some dedicated power supply upgrades will make for a really high performance stereo mic pre for much less than is possible in the music stores. that’s for the next installment.

and finally, i often left little secret messages on the PC boards. political ones mainly, as Bush was president. matthews is a rabid republican. on this one, there’s a little plug for the band i had at the time… elvisbeatlesgod. most people will never see those little details…

here’s a a few pix of us at SINE. that’s me and gary poulson in the background. gary was actually the drummer! which is nuts because he’s an amazing guitarist. way beyond me. yes, there’s a fair amount of electro harmonix on that stage, as well as a bunch of home made things too! most of it over on my side! but masako kawaii, our singer, used the prototype 12AY7 mic pre at nearly all our gigs. together with a 16 sec digital delay. both maiko endo and jim mckay, the strings, used EH stuff. maiko has a POG, a memory man and a white finger. jim uses a bass micro synth, a TUB EQ (as a wahwah)  and a memory man.

dorit chrysler in malmö

starting out is rarely simple or pain free. for me, it’s a messy unpleasant series of clumsy efforts and hesitation. one step forward, many steps backwards. throw it in the crapper and start over. i swear. i despair. i think dark bloody thoughts. i sometimes throw things. i do not feel proud of myself during this time, or happy. i know self hatred and shame. it’s a hole filled with bad. and yet, i have no choice but to go in there… an idea, a seed… a baby? a monster? like moths drawn to the porch light. i have to look in there. in the morning, the doorstep is littered with little carcasses. if this isn’t you, bless you… and screw off.  if you identify, read on brothers and sisters.

those who push the myth of creative genius do us no favors with their romantic fantasy. it’s good marketing, but mostly a pack of lies. making new stuff is rarely sweet. important? yes. satisfying? eventually. fraught with suffering? is the pope catholic? do bears shit in the woods? i’m not talking about doing the same old shit over and over again. it’s when you push for something new, different, untested, untried… but especially something that requires that you change your mind. i don’t think it matters if it’s an idea or a new material you just wonder about. just that it is not what you already know.

it’s good to go back to the reference for help with this: to remind ourselves what it is like giving birth! i’m a man. i have only watched and tried to stay calm. you do what you can… it’s not about you. at the risk of diminishing the magnitude of pushing out a baby (no man will ever know) it is still a good way to lend some understanding for what is going on in creative work in general. we are all mothers when it comes to this. the pressure is immense. it’s like crapping so hard, your bones turn inside out. and, you are messing with the contents of the universe. other people have strong feelings about that: “oh great, another one?!” or, “that’s god’s work you’re doing there, hallelujah!”. expectations… total indifference. there’s responsibility involved. it’s rough physically, even with the drugs, and there is blood and guts and risk. if it all comes together, there you are! with a new asshole and only at the beginning. and then the hard work starts. whatever you’ve made, becomes less and less yours the more complete it becomes. those really special things, the ones that end up as if they were always there long before you ever were and right and whole, those seem to have the least to do with you of all. even though all you’ve done is work. at least for a hacker, artist, or crafts person, there might be sleep at the end of the day, and not so many poopy diapers…

why am i talking about this? ahh. i was thinking about beginnings. starting out can be really hard. the middle usually ain’t so bad at all, and then finishing can get tough all over again. it’s a matter of dealing with the transitions… each one, a beginning. just as when kids are confronted with the switch from play time to lunch. or lunch to nap time. they often need a little help…

for recorded music, the start is often very very far from the end.

it’s interesting to consider how different the worlds of the recording engineer and the listener are generally… how little the two planets have in common, although somehow, they must be related?! the stuff looks similar! it relies on the same physics. the microphone and the speaker are reciprocals of one another, and definitely have many connections between them, both from a technological perspective and as a physical experience. speakers can often also be used as microphones too (very nice to use a 15″ bass cabinet as a kick drum mic, setup right in front of the kick). but, the one is at the beginning, and the other at the end. making music and recording it is anything but linear or neutral. playing it back is often practically puritanical. and there is a host of assholes wagging their fingers and tongues if you forget. the recording engineer makes a living with microphones, and the audiophile throws money at speakers… the recording studio is often a bland, foam lined suburban den looking crib. a fur lined ford fuck truck. the listening room is anything from couch potato modern to ikea tragic on ketamine. sometimes, the rooms are really carefully designed, other times, random chaos. ghetto or fly, they are obviously not ordinary spaces. one thing that is different now, than from years past, is that the site of the recording has become highly specialized. it is no longer a place most people will ever go to or hear any music. in many of them, the music doesn’t actually sound very good in the space at all! isolation is a priority in these spaces. it is a sort of temple, in the old sense. a space cut away from the ordinary world… just for the purpose of recording sounds. it didn’t used to be that way. music was recorded in places that you might ordinarily hear music: clubs, concert halls or temples! these days it is not at all like that. on the far end of the opposite side of the playback world, the listening room is actually a more open and inviting environment.

and then, there is how mics and speakers point at their various targets. that is also very different usually. i have spent a good deal of time trying to make speakers with a cardioid dispersion over the important octaves, just so i can set up a situation where the speakers really ARE the reciprocal of the microphones (stereo) in a really good setup. of course, only certain recordings can be ideally enjoyed this way, because modern recording practices seldom resort to such life-like techniques anymore (blumlein, ortf, decca tree, mid/side, etc). audio verite is out. interestingly enough, speakers which hold to this standard often sound great anyway… because the ears are prepared to hear things that way (a long conversation for another day). you don’t need to live in an anechoic chamber to enjoy the benefits. in any case, obviously, what happens at the microphone, the beginning of the recording, has a profound influence on the final result, regardless of the level and sophistication of post-production, and whatever you use to play it back. and, assuming the speakers are pointed at a listener (which they aren’t always…) some understanding of microphones and recording techniques help in speaker set up… although you NEVER hear that uttered in audiophile circles.

so what goes into a great microphone? what makes a great mic preamp? what makes for an appropriate mic setup?

good questions and not so easy to answer simply… have to deal with the first question first. there are several different kinds of mics and they lend themselves to different applications. and of course, different manufacturers handle the technology in varying ways. recording engineers like to think of themselves as really knowledgeable about mics but it is rarely so. they know applications… and are good at getting results using them. that is not the same thing. a few really do know how the gear works, but the vast majority get a good sound by any means necessary and stick with it, come hell or high water! not a bad idea when your paycheck is based on it. “who cares how it works?” quite often, the folklore of various mic types and the ways to use them are rooted in economic success, “hit sounds”, and that is the compelling standard. in american pop music, for example, recording a snare drum is most often done with a Shure SM57, because it has featured on most of the top of the charts recordings. everyone knows that sound and it causes no ambiguity. its also simple and repeatable. many people really do like that sound also, but most don’t question it. “that’s how the stones did it”, etc. i think it pays to think for yourself and keep an open mind here. carrots are delicious! i don’t eat them every day… still, i haven’t won any grammys! apparently, you should eat carrots every day. but, this isn’t the only situation that creates a particular niche for a mic though. the difficulty of recording certain sources in a really satisfying way also demands particular qualities and performance. piano, for example, is particularly tricky because of the distributed nature of the instrument. there are endless discussions as to what the best way is to mic a piano. there is clearly no one way. in this case, it is the opposite of the snare thing… certain engineers get known for their way of doing it, and also for the mics they choose to use. these two factors play most heavily into how and what is used, when. “steve albini does it that way…”, etc. a general overview is worth the effort.

the ear is sensitive to both differences in velocity and in pressure, and the universe has been generous to us in allowing ways for both types of variations to be detected and transformed into electrical energy. the condenser mic, and the dynamic mic are both efficient pressure transducers, and to a much lesser degree, velocity. ribbon mics, in their classic arrangement, are excellent velocity mics, and to a much lesser degree, pressure. at least, not so great as frequency goes up. there is a lot of overlap and exception in use but pressure mics tend to emphasize presence, and velocity mics tend to emphasize space. the distance to the source is also a factor, with proximity increasing pressure but not so much velocity. this will boost low frequencies as the mic is pushed closer to the source. recording spaces and recording particular sounds are very different jobs. the mic you choose for the job has a profound effect on the perceived result.

noise is always an issue and if you record noisy, noisy it will forever be. so impedance is an issue too. ribbon microphones are the lowest impedance and are theoretically quietest… although that is sure to find disagreement in certain quarters. typical range is 50 to 250 ohms after the transformer. before the transformer is milliOhms. it is just a short piece of foil after all. they put out such small signal levels (similar to a MC phono cartridge) and need a lot of gain after… the following passives and electronics play a big part in how well they do. dynamic mics are usually medium Z and have some meaningful inherent noise. the range is 250 Ohms to 5K Ohms. SM57s, for example, have enough self noise they cannot be used to mic a quiet source without obviously adding it to the signal. they hiss. still, dynamic mics can be medium to high output and that is also a consideration because the signal to noise ratio can still rate very well. there are some interesting low Z dynamics that are very quiet (Sennheiser notably). condenser mic capsules are inherently very Hi Z: on the order of 10 to 50 MegOhms.  so they also must be buffered. their noise is both connected to the impedance and the electronics required to extract the signal… they have the highest “self noise”, but at their best are still very quiet (Shoeps, B&K/DPA, Neumann and Sennheiser). because of the electronics they can have the highest output and very impressive signal to noise ratios. condenser mics require a polarizing voltage to charge the capacitor capsules and this can come from an external supply (called a phantom supply or a dedicated supply for tube buffered condensers) or internal (a battery). a trend in modern mics is to derive the power for the preamp electronics from the phantom supply. this means that these mics draw more from the supply than in the past, which simply charged the capsule. they are also very high output. many of the most well known and beloved condenser mics are not that quiet. if you are recording something close up and loud, who cares? if you are doing a large room with subtle crap going on, you really do. nearly all mics are also connected in a way as to avoid returning the signal on ground (balanced output), which shares the electrical environment with everything else in the universe. by floating the signal off ground, a large source of noise is avoided. this does require somewhat more complex electronics but you will rarely see an RCA plug on a mic in a recording studio. RCA plugs suck anyway.

presence and clarity are more important in contemporary recording than space and ambient detail, even in classical music… so putting the mics up close is an inevitable fact of now. it is interesting that this is nearly always the case today, and it is absolutely antagonistic to the notion of “natural” recording. this is one of the more obvious reasons why “natural” (it is electronic after all…) is no longer at issue at all. there is some honesty in this but honesty isn’t the motivation. in reality, we rarely listen to anything with our ear right next to it… at least not for long. can you imagine a trumpet or a kick drum in your ear? a whisper or a tongue, sure… not a trumpet. yet, it is common practice to stick mics inside of these. no it doesn’t sound anything like a trumpet or a drum in a room, or a hall, but it has it’s own sound that is taken for granted today. the engineer then “assembles” the separately recorded sounds into a “mix” that balances the various levels into something that makes aesthetic sense. no, it has nothing to do with the way things actually sound. but it has what the engineer or producer considered to be important to the feel and mood of the work. when space and ambient info is wanted, it is often added artificially with reverb, “room mics” (it may not be the original room at all, but a side room, or a special built “echo chamber”), or delay effects (tape delay, digital delay, or “analog” delay (bucket brigade chips or magnetic oil)). creative control is the concern here, rather than documenting an event. close placement puts demands on the mics in terms of overload and durability that are important features. quite often, at high pressure levels the mics put out considerable amounts of signal, and distortion. this is part of the deal too.

the dispersion characteristics of the mics are also various. there is omni, cardioid, and hyper cardioid. figure eight mics are dipoles… and the dispersion characteristics are created by the dimensions of the mic diaphragm, baffle size, and the electronics. sound familiar? the cardioid dispersion is the most common and generally useful arrangement. it can be used close up or far back. it is well suited for imitating the experience of two ears on either side of a head, that separates them. if you are recording larger or more diffuse subjects, such as a piano, or an orchestra, omni mics come to the rescue. they pick up EVERYTHING. and if you need to isolate one thing, in a crowded environment, such as a snare drum in a kit, a hyper cardioid mic is dandy. shotgun mics are hyper hyper cardioid and are often used in film making to isolate the subject away from the equipment and people necessary to make the film. again, this is a rough general description and in no way conclusive as to application.

my motivation for relating these issues is mainly for those who are primarily focused on playback: the design of amps and speakers. it is rare among audiophiles and even DIY hackers that they have even the slightest clue as to how the object of their affection is made. in the next installment, we will look at mic preamps. and i will give a practical example.

this will be a quick post about something i’ve been thinking about… been working on phono stuff for a while and i do love the sound of nickel LCR RIAA. i have to say that slagle has shown me the light in this regard. and i also do really like the “medium impedance” thing too… 7.5K is a fabulous compromise, easy to drive and being able to use standard value caps in the filter is the only way to go for me from now on. i hate paralleling all those little styrenes… however, it is not always possible to convince a customer who has their head wrapped around the pultec or emt/neumann preamp idea and a 600 ohm RIAA. what do you do? it is really asking a lot for a small signal tube to do that job. you need a 60 ohm output impedance to do it right and it needs to be able to pop and crackle sublimely… otherwise it is just a lot of hard work for nothing.

as many of you know, i am a hopeless contrarian. so i love hybrid tube/solidstate… but not the dumbass tube front end with a solid state buffer out. boring and not usually so fabulous sounding. although i really love JFET buffers. i just haven’t completely loved the mosfet buffer thing yet. that could change? nelson pass’ stuff isn’t bad at all… i need more time to try it out myself, to get used to the sound. it is really different. off hand, i think it’s a bit plain for my taste. and a bit hard. but not like halcro! yuck. no, for me, i like sneaky mixes. and i love that i can say fuck off to both the vintage crew and the high end guys at the same time with the same gadget! efficiency! that’s good engineering.

philips made some voltage references for power regulators in the 60′s that used a tube/transistor feedback cathode follower to get a lower output impedance and better stability. in an otherwise completely standard regulator circuit, with an EL500 as the pass element, they have a dual diff DC error amp with a buffered Vref on the first grid. the error signal on the other side. this circuit used the grid/cathode voltage of a cathode follower to sneak in a PNP transistor that amplified the output signal and returned it to the input. a 2 ohm Z out is claimed (for the Vref) using one section of an E80CC. interesting circuit. i am sure the reg works!  but i was more interested in the cathode follower. i have played with variations on this idea on and off… it isn’t so complex that it adds so many more parts, but is more complicated. the performance is excellent and the sound is there. but it puts limitations on the range of operation much as any feedback circuit. clip it, and things are often worse than they would be without the fancy tricks. the limit is on the input level, because of the feedback amplifier. up until it’s diode line, it helps… after that, it un-helps. the question is, within range, is it helpful driving a 600 ohm RIAA into overload and recovering fast? it has led me into some fun experiments with local active feedback around a cathode follower… something i think is a very interesting technique. it does allow a 5687 to drive the hell out of a 600 ohm LCR filter, something it doesn’t do so lovely without help.

in the schematic above you see the basic idea… the circuit on the left is it, i added the source coupled stage instead of a VR tube… (this is a buffer, not a voltage reference!) the middle is a classic CF. and the right hand side is something more out there. the classic circuit gets into trouble already at 5V PP into 600 ohms. pops and clicks from LP’s are ugly. the modified philips style is even more sensitive, although up until it does clip (about 7 volts), it is exquisite. it’s the JFET source follower that craps out. increasing the current increases the range, but things get hot. it does recover fast, though. the feedback helps restores things. the insertion loss is also very low with the FB buffer, something like .05dB (!). it’s 3 dB with the classic CF. the mosfet circuit has an extra local FB loop from drain to gate and it does get you something extra for the trouble. the distortion actually goes down (!) as the signal level increases. very interesting in a DC combination with a Gm amp and a LOT of NFB. that’s not for the DIY guys though…

2 VPP input, .04% dist for all into 600 ohms

10 VPP input, red= 3.8% dist, blue=2.7% dist, green=.02% dist

clicks and pops are so benignly handled by both FB buffers shown here, as long as they are clipped. they can be used at the output of a high gain block, and then drive the 600 ohm LCR at the output of the phono preamp as the load. simple, elegant and can save a low Z stage earlier in the pre. here’s the poop: the red trace is the CF, the blue is the philips style. and the green is the mosfet FB circuit. interesting stuff. if you don’t mind throwing current at everything, the input range can be improved quite a bit… still it’s the idea that inspires me.

here’s a quick update: i have rearranged the first buffer to draw more current at the source follower and the feedback amp… also, the load for the 4403 comes off the low B-. now the simpler circuit works as well as the more complex one. this one is also working with an input of 10 VPP.