Hi Jesse, Mervin,
This is one helluva question, and probably goes to the core of high end design.
As Bill Shakespeare, the high end audio linguist, once remarked, 'To buffer or not to buffer, that is the question.....'
1- would (in your opinion) a circuit this simple (signal only passes through ONE component more than a purely passive design (though granted, it is an IC)) benefit more from keeping the # of parts to a minimum, and keeping the signal path short, or from the addition of buffering of some kind on the input?
Jesse, you actually address two pivotal issues here. Do we use a buffer on the input, and do we use an IC? There are two facts to consider. Firstly, we know interconnects have quite considerable impact on sonics, and as modern equipment steadily improves, many of us notice this more and more. This should not be so (and has spawned religious cults!), but clearly is, and relates to parasitic reactances, mostly capacitance but also inductance, and attendant effects on the following stage, often tied up with short term stability. RF technology calls attention to the issue of 'terminating impedance', which becomes important at around 500KHz, which happens to be close to the primary pole on most audio equipment using global negative feedback, and which might thus be expected to have cascading effects down into the audio band. All my work with lag compensation on voltage amplifiers tells me that anything which impacts on this pole frequency, and hence stability, is immediately audible. But that's another story.......
Secondly, sources are not perfect, and like software, suffer from a myriad of standards, many of which are ancient, irrational, and just plain wrong. (The design of the RCA plug, for one.) Thus source impedances vary markedly, and so do the phase margins of their output stages. Stability, as commented earlier, is profoundly affected by terminating reactances.
On consideration, and given that pots tend to be chock full of parasitics (which partly explains why there is such huge sonic variance from pot to pot), it seems logical to use a buffer before the attentuation function. This buffer would need highish input impedance (not too high because this makes the interconnect highly susceptible to hum intrusion), high dynamic range, minimal intermodulation, restricted bandwidth to remove RF which might have injected itself into the interconnect, and vanishingly low output impedance. Clearly it does not need gain, so an opamp need not be used. That's some relief, I guess!
This leads you to the buffer IC, which is relatively cheap, but which generally uses at least three cascaded emitter followers, and sometimes even a voltage gain stage, throttled back to unity with nfb. As Mervin has found, these are damn good for all their complexity, but their output stages are normally Class AB emitter followers in push pull, with all the problems found in conventional amps along these lines and without the powerful charge suckout technology employed in the better SS amps to eradicate the crossover disjunction. So, this deficiency leads to my next point.
The conflicting requirements of very low output impedance, some bandwidth limitation, single ended class A circuitry throughout and highish input impedance leads inevitably to a differential pair input, a conventional, carefully selected transistor with very low parasitics for a voltage amplifier operating in constant current, and a global negative feedback loop. Like it or not, the Bailey configuration used in the AKSA is one of the oldest and best configurations for audio that has ever been discovered. So, we can use it as a buffer, and by dint of nfb extract vanishingly low Zout. A current source can drive the voltage amp, and since we have Bode/Nyquist stability criteria to meet with any global feedback arrangement, it will be inherently band-limited to around 50KHz.
Another boon to this configuration is low parts count, since just two transistors effectively constitute the signal path. That's pretty cool, and impossible to beat with an IC, since the manufacturers appear to feel a powerful obligation to use at least five active devices in the signal chain, and this, IMHO, often compromises things sonically.
Jesse, I believe the foregoing answers your questions in para 2.
3- what are your general thoughts on this particular design? here is a link to the schematic minus the PSU which i have modified and don't have the design handy, but suffice to say that it will be dual mono, and more than is probably necessary for this IC, but i want to make sure that i'm getting ALL i can out of this lit'l buffer.
The design would work fine, but would be quite susceptible to quality variations in the interconnects. It might also load down the source beyond its comfort zone, since you say you'd like to use 10K pots, or even lower, and while CD players have a Zout around 100R, very few are comfortable with 10K loads. And the vast majority of tube designs are also uncomfortable with loads this low; from decades ago most tube amps used a Zin of 47K to give the tube pre an easy life.
I hope this gives you a brief tour of the management of compromise which, like most technologies, dominates high end!
Cheers,
Hugh
