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
Hi all,

It's great to get some brilliant designers involved in this thread (ain't sucking up ... I call 'em as I see 'em).

Whoah, Thom! Look what you started here.

I'm loving this discussion and hope that it will bring light to the general philosophy of how we achieve the goal of a realistic and SATISFYING musical illusion in our homes. I hope this is everyone's goal. Yes, I'm 5 small steps ahead of Doug in this area and miles behind you guys, and that's part of the fun in watching this thread unfold.

Indeed, the automobile metaphor was incomplete at best, but you need to kick off a conversation with a provocative statement. Like all designs, it is the entire system context that is important, as Nick, Jim, Ralph, Raul, and Jose all pointed out.

Jose brought up some great points, but I'm still puzzling through the difference between what Arthur Salvatore would call "noise floor" and "sound floor". My Artemis is not the quietest piece of gear out there (noise floor - it is quiet but not the quietest) and yet it has the uncanny ability to extract musically significant nuance from recordings (sound floor). Other great designs accomplish this differently

I too have read from more than one individual that a phono cartridge is not balanced, but rather floating single ended. Victor Khomenko's (BAT) writings over on Asylum are one source, and I've heard Jim comment about this on several occasions.

If I recall correctly, when you "balance" a phono cartridge (using a center tapped transformer, for example), you don't get the normal 6dB common mode rejection, because you are "halving" the signal in order to balance it. My explanation may not be perfectly clear on this.

I too have to wonder about the necessity of obsessing over small fractions of a dB in RIAA accuracy when in room speaker response varies by such huge amounts. I would agree with Jim however about left vs. right channel balance as far as treatment of the delicate stereo signal and its implications for imaging and such.

Now, I certainly understand that distortions build through the signal chain or through the gain stages in a single component for that matter - that if you were forced to choose in reducing distortions within an active amplification stage, that larger gains would be had by "improving" the earliest gain stage.

Surely you want to address all issues if you can, but the point here relates to the impact cleaning up the signal as early as possible. How much is too much, and when should we shift our focus away from trying to get the RIAA eq. to vanishingly low levels? I suppose the only way to validate this would be to construct an experiment and introduce larger and larger RIAA errors in several different systems.

Cheers,
Thom @ Galibier
Ok- I have to clear up some misconceptions:

To have true balanced, you need 3 signals. A reference level (return), positive polarity, and negative polarity. A cartridge obviously has only 2 pins per channel.

The reality is, the pins of the cartridge are the inverting and non-inverting outputs of the cartridge. In a differential balanced system, there is no ground signal at all- ground exists only for shielding and it is possible to operate a balanced line without shielding (IOW with only 2 wires...). In this case, the ground wire is the shield and is also the tone arm itself which is shielding the wires.

>>RIAA eq deviation no more than 0.05db<<

Why?

Because if there is an error, it won't be *our* error. You'd be surprised how good the pre-emphasis on older cutting machines actually is. My Westerex system employs hand-picked components to insure accuracy against the serial number of my cutter head.

If I recall correctly, when you "balance" a phono cartridge (using a center tapped transformer, for example), you don't get the normal 6dB common mode rejection, because you are "halving" the signal in order to balance it.

The balanced source which is the cartridge arrives at the input of the balanced phono section without any transformer. If you refer to my comments above, the inverted ('-' pin) goes to the inverting input (pin 3 of the XLR), the non-inverted output ('+' pin) goes to pin 2 of the XLR and the tone arm ground to pin 1 of the XLR. Very easy, very simple, and nothing 'halved'. The cartridge is normally a balanced source and you get common mode noise rejection like crazy.
Dear Thom: +++++ " I too have to wonder about the necessity of obsessing over small fractions of a dB in RIAA accuracy when in room speaker response varies by such huge amounts... " +++++

The proof is is on the hearing and you can heard it through the Essential 3150, as a fact we heard at your place.

+++++ " experiment and introduce larger and larger RIAA errors ... " +++++

We already do it ( in some ways and not in perfect way ) and that is why I speak in absolute terms about.

Again, your quality sound reproduction targets/priorities ( like JH ) are a little different from ours: we are looking for the perfect sound reproduction, perhaps we never achieve that target but in all ways we will work hard to be the nearest that we can and the inverse RIAA eq. accuracy is one of the targets that help to achieve that very high target.
We can't tolerate distortions/colorations/noises everywhere and we can't accept ( in any way ) that because the speaker/room interaction produce big frequency deviation then we don't take seriously what happen in other links in the audio chain taking in count when we have the control and the knowledge for lower that distoritons/colorations/noises in the Phonolinepreamp and in the future in the amplifier and tonearm.
Thom sorry but I have to disagree with you on that subject: every single " sand grain " is important, the inverse RIAA eq. accuracy in our Essential 3150 is only one way to lower the colorations in our design we have several other subjects that help too to lower those colorations/distortions/noises.

Regards and enjoy the music.
Raul.
Thanks Ralph (about balanced) ...

Yours is a different perspective from the limited commentary I've read on the subject. I'm not trying to be contentious, as I'm clawing to grasp some of the subtleties here. A while ago, I dug up and filed away a few links to threads on the topic of balanced. Some on the list may be interested in reading more, so I'll paste these in as hyperlinks.

Asylum Thread - on Balanced Phono
Asylum Thread - another Balanced Phono thread
Asylum Thread - on the noise disadvantage of running phono as a balanced device

Hi Raul,

I'm confident that everyone who is active on this thread has exactly the same sonic goals as you do - goals which you share with your partner Jose but which he stated in a way that I think more of us can relate to. I think we're working our way through a language issue here.

I think that your comments about searching for perfection is another way of your saying that every little bit helps (as Ralph agrees).

Now, in the case of a passive RIAA circuit, achieving low variance is more a matter of painstaking attention to parts matching, along with possibly cost, as you end up having to reject R-C pairs which don't result in a correct time constant (turnover frequency) within your specifications.

This is a time and expense sort of thing and not a design challenge. From a design perspective, it's a "freebie". Correct me if I'm wrong.

OTOH, design approaches have the potential to result in sonic penalties elsewhere in the design.

Perfection is a noble goal. It is however like saying you want to stop world hunger. Everyone interested in stopping world hunger raise your hands now. It becomes meaningless to the point of sounding like marketing.

Back to design traps, I think you'll agree with me when I state that the designer can easily paint himself into a corner by trying to track down microscopic distortion levels. Here's what I mean. The goal of lowest distortion might involve the implementation of feedback somewhere in the circuit (still using passive RIAA equalization, which seems almost universally accepted). One might be tempted to dial up the feedback until measured distortion is minimized. Overall "performance" might drop at this reading however. I'm reminded of a Tom Robins rant on the word "Performance" ... don't get me started.

Chasing down noise is another one of those demons which can get you into loads of trouble. In one of the above Asylum threads, Victor Khomenko (BAT) comments:
"Contrary to some beliefs, the signature of a good circuit is NOT no noise, but it is GOOD noise."
You can read his analysis in that thread, but I find this to be a provocative statement worth quoting, as it has great implications to how someone might approach a design.

I'm sure you will agree that the key is to think of the entire design holistically. The results in your Essential bear out that you embrace this philosophy. You don't get lucky with such a complex design. Please however let's not turn this into another thread on the Essential. We have an active thread for that.

BTW, you throw specs around. I'd be curious about how do you measure distortion - with sine wave input? with a wave form of an orchestra going full out?

Cheers,
Thom @ Galibier
"RIAA eq deviation no more than 0.05 dB... Why?"

Five reasons:

1. Because the technology exists
2. We can use this technology without any detrimental effect on the audio quality
3. It adds a negligible cost to the product
4. It automatically results in basically perfect channel balance throughout the full audio band
5. It'd be a tribute to Dr. S. Lipshitz (of RIAA eq. fame)

Of course, RIAA eq. is only one of many parameters affecting the reproduction of LP records. I support that. As for channel matching, please see point 4 above.

"What's next? +/- 0.00005 dB equalization?"

That's impossible!