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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
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atmasphere's avatar
atmasphere
11,894 posts
 
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.
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rauliruegas
13,103 posts
 
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.
thom_at_galibier_design's avatar
thom_at_galibier_design OP
729 posts
 
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
jmaldonado
57 posts
 
"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!
jcarr
382 posts
 
The magnitude of RIAA error is not particularly useful unless we also consider the range of frequencies that are affected by the error. in practice, a 1dB deviation that only hits one note is not going to be very noticeable, but a 0.1dB error that spans an octave or more can be quite noticeable. In fact, the manner of musical presentation changes when this happens. That said, I do think that the less total deviation there is from the RIAA curve, the better. True, LP recordings and mastering systems have their own deviations, but they can deviate in any direction, and as long as we keep as close to the standard curve as we can, the frequency deviations in one's LP collection should average out. At least we won't be favoring certain recordings over others, which would certainly be the case with an RIAA playback network that wasn't right.

I don't think that speaker colorations are an acceptable excuse to tolerate RIAA deviations. Admittedly it is next to impossible to exactly duplicate electronic colorations in the speaker and vice versa so that they can be truly compared, but at least in my experience, it has seemed that electronic colorations are much more noticeable and less forgivable than speaker colorations. I think that this is because acoustic colorations in the environment are part of everyday life, and compensating for this is a constant, subconscious process.

Regarding when balanced phono amps were introduced, I have on my bench a schematic for a discrete FET balanced phono amp from the Japanese audio magazine M&J which is dated January 1985, and I am pretty sure that there are earlier examples (especially from the tube guys).

I don't think that complexity in a design is necessarily a bad thing, because a major goal of this approach should be to get smaller "modules" with more well-defined tasks/behaviour/environment. This makes it easier to design, understand and debug the functions and can lead to better performance, even if the overall complexity becomes greater. A simple circuit can lead to a wider, less clearly defined range of responsibilities being assigned to fewer parts, and this can result in lower performance.

IME, NFB is just another tool, neither good nor bad by itself. The results of using NFB have a lot more to do with the capabilities and sensibiities of the designer than NFB per se. I usually dial in the amount of NFB by ear as well as by measurement, and sometimes I'm at 0dB of NFB, sometimes 50dB.

Although I fully agree with the "holistic" approach, I think it is possible to achieve a good-sounding line or phono amp using a variety of technologies, circuits and approaches. However, topologies and components by themselves don't know what they are supposed to sound like. Good sound, bad sound, they don't know any better. The most important component of all is the designer, and the final sound extracted from the topologies and components is only as good as the designer allows.

regards, jonathan carr
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