What Makes a Good RIAA or Line Stage?


Hi Doug,

In a currently running thread on a certain RIAA / Line stage beginning with the letter "E", some very provocative comments were made that are of a general nature.

I fear that this conversation will be lost on the many individuals who have soured on the direction which that particular thread has taken. For the purpose of future searches of this archive, those interested in the "E" thread can click this link.

For the rest of us who are interested in some of the meta concepts involved in RIAA and Line Level circuits, I've kicked this thread off - rather than to hijack that other one. In that thread, you (Doug) mused about the differences between your Alap and Dan's Rhea/Calypso:

... the Alaap has the best power supplies I've heard in any tube preamp. This is (in my admittedly unqualified opinion) a major reason why it outplayed Dan's Rhea/Calypso, which sounded starved at dynamic peaks by comparison.

Knowing only a bit more than you, Doug, I too would bet the farm on Nick's p-s design being "better", but know here that "better" is a very open ended term. I'd love to hear Nick's comments (or Jim Hagerman's - who surfs this forum) on this topic, so I'll instigate a bit with some thoughts of my own. Perhaps we can gain some insight.

----

Power supplies are a lot like automobile engines - you have two basic categories:

1. The low revving, high torque variety, characteristic of the American muscle car and espoused by many s-s designers in the world of audio.

2. The high revving, low torque variety characteristic of double overhead cam, 4 valves per cylinder - typically espoused by the single-ended / horn crowd.

Now, just as in autos, each architecture has its own particular advantage, and we truly have a continuum from one extreme to the other..

Large, high-capacitance supplies (category 1) tend to go on forever, but when they run out of gas, it's a sorry sight. Smaller capacitance supplies (category 2) recharge more quickly - being more responsive to musical transients, but will run out of steam during extended, peak demands.

In my humble opinion, your Alap convinced Dan to get out his checkbook in part because of the balance that Nick struck between these two competing goals (an elegant balance), but also because of a design philosophy that actually took music into account.

Too many engineers lose sight of music.

Take this as one man's opinion and nothing more, but when I opened the lid on the dual mono p-s chassis of my friend's Aesthetix Io, my eyes popped out. I could scarcely believe the site of all of those 12AX7 tubes serving as voltage regulators - each one of them having their own 3-pin regulators (e.g. LM317, etc.) to run their filaments.

Please understand that my mention of the Aesthetix is anecdotal, as there are quite a few designs highly regarded designs which embody this approach. It's not my intent to single them out, but is rather a data point in the matrix of my experience.

I was fairly much an electronics design newbie at the time, and I was still piecing my reality together - specifically that design challenges become exponentially more difficult when you introduce too many variables (parts). Another thing I was in the process of learning is that you can over-filter a power supply.

Too much "muscle" in a power supply (as with people), means too little grace, speed, and flexibility.

If I had the skill that Jim Hagerman, Nick Doshi, or John Atwood have, then my design goal would be the athletic equivalent of a Bruce Lee - nimble, lightning quick and unfazed by any musical passage you could throw at it.

In contrast, many of the designs from the big boys remind me of offensive linemen in the National Football League. They do fine with heavy loads, and that's about it.

One has to wonder why someone would complicate matters to such an extent. Surely, they consider the results to be worth it, and many people whom I like and respect consider the results of designs espousing this philosophy of complexity to be an effort that achieves musical goals.

I would be the last person to dictate tastes in hi-fi - other than ask them to focus on the following two considerations:

1. Does this component give me insight into the musical intent of the performer? Does it help me make more "sense" out of things?

2. Will this component help me to enjoy EVERY SINGLE ONE of my recordings, and not just my audiophile recordings?

All other considerations are about sound effects and not music.

Cheers,
Thom @ Galibier
128x128thom_at_galibier_design
>>is frequency response at the phono stage is magnified by the time it reaches the speakers?<<

No it isn't. Deviations add up, but they don't multiply. A 1dB phono error will give you a 1dB shift in frequency response at the speaker.

How much is that? Well, take your listening position. Now move your head about 6 inches in any direction (up, down, front, back, left, right). That change you hear is probably more than 1dB.

>>A deviation of 0.1dB is I think 1% accurate<<

Yep. Here's a plot I made years ago showing what happens when the capacitors are off by +/-5%. You get peak errors of about +/-0.4dB. I'm sure the same sort of thing happens with resistor tolerance.

www.hagtech.com/images/accuracy.gif

>>Johnothan's calcs re the effect of cart resonance<<

He got it right. Resonances can be ultrasonic.

www.hagtech.com/loading.html

More info on RIAA at:

www.hagtech.com/pdf/riaa.pdf
www.hagtech.com/equalization.html
www.kabusa.com.riaa.htm

jh
Dear Bob: In a " perfect system with perfect audio devices that 1db deviation or any deviation that comes from the phono stage will be the same at the speaker output, but there is no perfect audio devices.

A signal that comes from a phono stage has to pass through: cables/connectors to the line stage, then in to the line stage, again to cables/connectors to the amplifier, then inside the amplifier and through the speaker cables: in all of these single " stages " the signal is suffering a degradation over the deviations that already has the signal that comes from the phono stage, when this signal goes out the speakers, IMHO, those deviations will almost be magnified but only with measurements about we could know for sure how much.

Regards and enjoy the music.
Raul.
>>3.18uS turnover point ... switchable ... other components altered at the same time<<

You are indeed correct. The 3.18us cannot simply be switched in and out. Adding the corner affects the other component values.

jh
Oops, I made a mistake. The above is true only for the standard Lipshitz method of EQ. If EQ is split across two stages (the way I do it, for example), then the switch is possible.

jh
Regarding the rationale for the non-standard 3.18uS (50kHz) turnover, first it needs to be established that all cutting lathes have their HF resonances in this same region. Is this true for each and every one of the Neumann models, and what about JVC, Sculley, Westrex et al and their respective models? Next, it should be pointed out that half-speed mastered LPs will have this HF resonance shifted by one octave (100kHz instead of 50kHz).

regards, jonathan carr