What Makes a Good RIAA or Line Stage?

Thom,

Sheesh, I'm glad you didn't TITLE your thread "Hi Doug". Opening with a salutation is embarassing enough. Still, I'm glad if my speculation on p-s "goodness" helped spark an interesting thread.

Once I heard the musical clarity of effective power supplies, it became much easier to hear and identify the musical mud caused by ineffective ones. Some are just inadequate. Others are overly complex in the way they're regulated. I don't have the knowledge to explain one vs. the next but I can hear the differences. Anyone could, with a little exposure.

In addition to a starved p-s, another sonic characteristic I've heard in many preamps is a kind of general congestion or confusion of phase timing and frequencies. It sounds (to me) like the result of overly complex designs and, as you already said, too many parts. Being an electrical ignoramus I can't put it any more accurately than that.

I've heard three preamps which don't do this (or rather, which do it notably less than all the others). They come much closer to Bruce Lee, which is a pretty good analogy for my ideal too.

Hope you get some responses from people who actually know what they're talking about. It's an interesting topic.

Doug

Hi Doug, Do you have any pictures of your system? I think it would help myself and others when we read your responses as to a reference to what you are listening to. Also what are you sound reproduction goals?
Bob

Dear Thom: IMHO the audio device whole design is what determine which kind of PS do you need.

Which are the voltages/current where the active parts run optimal for the best quality performance, overload margins, we can run those active devices full or at 30% of their specs ( like in the Essential 3150 ), what are the current/voltage needs in the worst stage, cascode source/fountain or not, type of regulation and filtering, how many ps stages, gain stages, needs on output level, distortion/noise levels, chokes or not, whole needs of the whole circuits, separate ps for each " stage " or not, circuit running in class A or what?, ps " power on " all the time or not, etc, etc.

As you can see there are more a lot more subjects than only capacitance. A good ps has to take in count those subjects and many more ( including the ideal capacitance number ) and the " secret " in a good ps design is to optimize any single ps area.

A good audio device design with a bad ps design will sounds bad and a good audio device design with a good ps design will be sound superb.

regards and enjoy the music.
Raul.

Hi Raul,

Indeed, holistic thinking is required - both in life and in hi-fi.

As you well know from how long it took you to get the Essential to this high level of performance, thinking holistically can drive you crazy as the circuit gets more complicated.

Not that the goal cannot be achieved, as you've demonstrated with the Essential, but the problem certainly becomes exponentially more challenging with each feature (or circuit block) you add.

I'd love to hear other designers' comments ...

To Doug ... a bit of embarrassment brings a rosy glow to your cheeks (grin).

Cheers,
Thom @ Galibier

Hey Doug, what kind of car should I drive???

O

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Atmasphere recently shared some interesting thoughts on the subject, you might want to check the archives.

Other designer's comments? Well, I think you guys got it right here in that power supply design is critical to optimal performance. It is often overlooked and/or not thought of as integral to a linestage of phono design.

I'm not familiar with the Alap, so cannot comment.

For me, I design a power supply as though it is part of the gain circuits. They are tied together and symbiotic in so many ways. Having said that, there is no single topology that works everywhere. A balanced differential stage works best with one type of supply, single-ended another. Opamps prefer something else. It is not about the amount of capacitance or inductance. It's about providing exactly what is needed for each particular case.

You also need to think of the box and system as a whole. That is, a box will have both low voltage and high voltage supplies; input circuits and output circuits. Each has to do their job without disrupting the other, yet play together as needed. And that box also has to integrate with other boxes and not cause them problems. A power amp should not mess up the linestage. So a good power supply thinks outside the box.

Hmmm. I seem to be saying exactly the same thing Raul did above.

The car engine analogy sort of leaves me short. To me, there are so many other parameters involved. It would be nice if it were so simple, but I can see how it helps to advance the discussion. Bruce Lee would indeed make a great power supply, if he could do so quietly without all the howling.

jh

Bob,

Thanks for your interest. I've been promising to post photos for years, so your nudge was well and truly due. Soon, soon...

I described our musical and sonic goals on our system page in this post:
http://forum.audiogon.com/cgi-bin/fr.pl?vevol&1051822874&openflup&65&4#65.
They haven't changed.

It's worth emphasizing that we are not big rock listeners. Many people find the resolution, transparency and neutrality of the sound we're seeking ill-suited to rock. We are constantly seeking ways to reduce artificially added "glue". Bloating or leading edge gentling from a component may help heavily mixed recordings sound more enjoyable, but they also make acoustic instruments and voices sound slow, congealed, larger than life and unnatural. Our favorite cartridge is a prime example. It positively refuses to add artificial overhang to any note. Natural decays are extended and reveal the sound space very clearly. Unnaturally bloated decays, which help blend the added feedback and multiple mixes of highly engineered recordings, are fairly non-existent. "AM radio" mixed recordings sound spectacularly awful in our system.

Like most tube preamps, ours can be tuned to provide more or less "glue" by rolling tubes and even (a little) with different isolation devices. Our personal choice is for the fastest, most neutral, most extended and least microphonic tubes we can find. This actually makes our preamp sound alot like Raul's on some music. They're closer than any other two preamps we've heard.

Others may prefer a different flavor of course. FWIW, we agree with Raul's longstanding contention that most tube components (and many vinyl systems generally) are tuned toward the warm and rounded. That is not our particular preference. We don't listen to sweet. We certainly don't post sweet. We don't even like sweet wine. It's pretty dry around here. If the sweetness isn't in the music we won't add any.

Doug

Nrchy,

I didn't know you owned a car. Once you've driven a Harley, what's the point?!

Hi Doug, Thanks for the response. I am looking forward to seeing pictures. It is important to understand ones sonic priorities as perhaps no ones are the same? Even though there are certain "threads" that hold them all together. By the way one does not "drive" a motorcycle, you ride them. The difference is that motorcycles have saddles, not seats. Well unless it is a Harley, in which case "driving" may be the correct word.

Bob

I have been smitten by Tom Evan's work for several years. His products, for me, epitomize the idea of simplicity. Many celebrate his decades work on the importance of a well-designed power supply. His black boxes are almost too humble looking to be respected in this hobby. The resulting atmosphere you find yourself in is nothing short of breathtaking. He answers both of your questions with a resounding yes. I am of course a total Evans junkie.

Hello,
my .02
1- there are no perfect supplies or audio amplifying circuits
2- Simpler is usually better
3- As far as possible, design the supply so that it really does not even care that there is a amplifier circuit attached to it, make its output impedance close to even across the frequency spectrum (as far as possible)
4- Design the circuit so that the change in current drawn from the supply is small so you reduce supply modulation
5- In the circuit design, pay close attention to the signal and supply ground for each stage, the loading of the preceding stage into the next and try to run devices conservatively enough for long life.

There is a large tendency to mythologise the whole aspect of audio design and it needs to be brought down to earth a little. It serves our egos well and may help sell a couple of preamps but lets get real, none of us was nominated for the peace prize because of our phono stage.

Thom has it totally right, does the thing get out of the way so you can understand what the composer/artist/singer was trying to do. The rest is BS. There is no supreme topology.
Even the person who brought up Tom Evans should realize that Tom started Acoustic Precison on the basis of audio circuits supplied by very high speed regulators and has now come full circle by using triple choke regulation in his latest amps.

I try my hardest to not post to the forums anymore because it always seems that someone either is spoiling for a flame war or has a dogmatic view that they wish to evangelize.
I think Thom is making an attempt to change the tone a little and I thank him.

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Thom, Great thread.
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Doug, Nice acceptance speech.
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Nick, Thoughful and appreciated post...I hope it resonates with a few people.
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Rgds,
Larry
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Thanks to you designers for posting your thoughts. Good one, Thom!

In my mind the next logical question would concern the RIAA curve itself. I'm sure we'd all agree that following the curve as close as possible is a good thing, but just how close does the implementation (not the design) have to be before the listener begins to have doubts about what he or she is hearing?

The idea that power supplies somehow compare to engines falls apart fairly easily. What if you have large capacitances *and* tight regulation for example?

In addition to excellent power supplies, a good RIAA section will be low noise with as few stages of gain as possible, to minimize distortion. In addition, the best of them will be tube, and passive EQ. Passive EQ allows one to avoid negative feedback, which puts an unnatural 'sheen' on top. Expect first-rate components throughout- handpicked to meet the RIAA curve and for low noise, etc. If on a PCB the material will be something other than FR4 to minimize dielectric effects of the board. Layout will include star grounding for low noise, and attention to RF suppression and improving stability by providing grid stop resistances and local power supply bypass.

I am also of the opinion that since the LP is recorded in balanced mode, and since the cartridge is balanced, that the phono section ought to be balanced too and to that effect we created the first balanced phono preamps in 1989.

The line stage ought to have similar characteristics- simple signal path, zero feedback, quality components, etc. A little understood issue is that the line section is supposed to control the interconnect cable to the point that the cable drops out of the system equation- IOW the cable between the amp and preamp will make little difference. If this is not the case for you your line section lacks this ability.

The volume control will be manual multi-position switch, or if it is remote, a motor drive operating a multi-position switch which is the only way to build up a decent volume control.

Power supplies will be outboard to minimize noise, and locally bypassed for the same reason. I can go on and on...

I cannot contribute to a technical discussion here, but can explain, anecdotally, what I have experienced in preamps and phonostages I have lived with since the early 1970's. My first serious foray into separates involved an old solid state MacIntosh preamp (I think it was a C-22) and a Dynaco 400 power amp. The amp had oodles of power at the time for driving dynamic speakers, but the preamp sounded compressed. I transitioned to a more modern solid state preamp by a 'fringe' company- Quintessence- which was more dynamic and quieter than the Mac, but had nowhere near the bloom or aliveness of the ARC SP-3, which I then switched to, along with tube amps and Quad ESLs. (Remember, we are still circa about 1974-75).
When the Sp-10 preamp was introduced, I eventually gave up my SP-3. Differences? Less euphonic, at least by comparision to the SP-10, and far more dynamic. Downsides- noise, tube anomalies, very sweet phono stage, and whatever bass or upper range limitations the preamp suffered were not readily apparent on the limited bandwidth system I was running.

Currently using a Steelhead with a Lamm L2. The Steelhead is dead quiet and hugely dynamic, but sounds a bit threadbare- I also am using the MM input rather than the transformer. With the Lamm linestage in the chain, the proceedings take on a richer, more involving, but less powerful presentation at the frequency extremes. I am probably willing to trade off some of the bandwidth for the extra life and body that the Lamm brings to the table.
With horn type speakers, I am also obviously less willing to tolerate noise.
One of the common things to both the SP-10 and the Lamm is a tubed-based power supply. I don't know if that contributes to the liveliness I like.

Whoah, Thom! Look what you started here.

I have to disagree with the assertion that a cartridge is balanced. It isn't. It's floating single-ended. 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.

However! You can turn force it into balanced mode by creating the reference (middle point) for it externally. This is most easily done using a step-up transformer that has a center tap on its primary. You ground the center tap and connect the cart across the entire primary. Ralph, I assume this is what you do? I hooked up a Trumpet phono once this way through XLR inputs. Works great.

The other alternative is to use an opamp type differential stage. Also called instrumentation amps. The drawback with this technique is a relatively weak dc impedance on the reference (permitting hum), unless you also drive it actively.

jh

A Line or RIAA stage could be characterized as "good" if it complies with its primary objective: To amplify the music signal with enough gain to be listened comfortably at the proper volume level, and do it without introducing obvious anomalies not present in the original recording. I think many of today's products could fit in that description.

Do you want more than merely "good"? First, the device should be accurate, transmitting all aspects of the reproduction with completeness and neutrality. With accuracy I'm not implying a clinical, analytic or sterile sound. This is not accuracy, is just, well, sterility. It also should be able to process even the most dynamic signals without any trace of compression or congestion. Every nuance, every detail, every "emotion" should come out in the right proportions. Nothing should be added or removed. The device should be "transparent" in the sense that listening to it would give you the feeling of being closer to the original event, not processed through electronic circuits.

As for the circuit details, there are probably as many different opinions as there are designers out there, but some general desirable characteristics could be extracted. It should have as extended a bandwidth as possible, both ABOVE and BELOW the audio band (officially 20 Hz to 20 kHz). Low noise is absolutely fundamental. A low distortion is highly desirable, because a high amount of it can be easily discernable as an added "gloss" in the instruments. Feedback can improve the measured specs, but if not properly done it will also rob life from the music (causing that sterile sound). In the RIAA department, it should provide clean gain for your cartridge (some of them needing up to 70 dB), and be able to manage input signals of at least 10 times without overloading. It should decode the RIAA curve with the minimum error possible, or it will show up as a permanent color of your sound. Finally, the unit should be reliable, stable, and have a fast warm-up time.

We now have the technology to satisfy all of these requirements simultaneously. If a unit aptly does that, there is a good chance that you will experiment all that "emotion" trapped in your recordings.

Jose, or anyone else, have you correlated RIAA error to colorations? As in, error of x amount or of y type leads to a z coloration.

Is that a clear question? :)

Dear Dan: Absolutely and José and I already tested about, the colorations/distortions/degradation are differents depending where in the frequency range are those inverse RIAA eq deviations.
Almost all phonolinepreamps have its own " colorations " other than the RIAA eq but the ones in the RIAA are more " be present at " because any single frequency deviation affects almost three octaves and this three octaves deviation we noted like a coloration, this special coloration is really a degradation to the cartridge signal because those frequency deviation does not exist in the cartridge signal.

One of the first critical and important task in any phonolinepreamp ( is one of the reasons why they exist ) is to mimic in accurate way the RIAA eq, this inverse RIAA eq must be that " mimic/accurate " to be ( at least in this stage ) truer to the recording .

A good phonolinepreamp IMHO has to have a inverse RIAA eq. deviation no more than 0.05db from 20 to 20kHz.

Our Phonolinepreamp RIAA eq deviation spec is: 0.015db but normally we are below that figure. Obviously that this spec not tell us how the audio device will sound, José already explain other very important subjects to have a good quality performance in a Phonolinepreamp device.

Hagtech, I agree with you about the misunderstood when we name " balanced " to a phono cartridge, it is not, our design is fully differential input to output and the connection is " floating ".

Regards and enjoy the music.
Raul.

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

Why?

Just how many cutting lathes in the 50's and 60's were built to this specification? And let's put this into perspective by comparing to the average speaker response. Why are speakers allowed 100x more error? Does that make sense?

The key here is that the stereo channels be matched. They need to be very tight. The relative matching is what preserves the spatial cues, soundsage, focus, etc. An absolute error of even 1dB can be reasonable as long as the channels are equal.

Hey, I've been promoting accurate RIAA eq for years, but let's not go off the deep end like the 70's chase for lower THD. What's next? +/-0.00005dB equalization? This is looking to me like its all about marketing.

jh

Dear Thom: About the line stage and specially in an analog audio system the whole design has to be very carefully and almost perfect in every single block on the line stage.

When we begin the Essential 3150 Phonolinepreamp design we don't start with the phono stage but with the line stage, one of the reasons was that the signal that comes from the phono cartridge not only has to pass through the phono stage but through the line stage too and at this line stage we have to take care that the signal does not " suffer " any or almost any degradation, from this point of view the line stage is critical at the same phono stage level design and that line stage has to be optimal in all circuit/operation parameters to conserve ( at least ) the same quality performance that comes from the phono stage, very hard to achieve but IMHO this has to be our target about. The people that already hear the Essential 3150 comments that the performance is good not only with analog source but with digital source too.

Regards and enjoy the music.
Raul.

Dear JH: I respect your point of view and I understand it.

My point of view is a little different: I'm looking for the best don't distorted/colored sound performance, the high deviations in the RIAA ( as you say ) have to be added to many other deviations in the whole audio system and this means that when the sound comes from the speakers what we are hearing is the original signal plus/with a exponential deviations/distortions added.
What happen if any one of us try to remove or at least take to a minimum all those frequency deviations/distortions in any single link in the audio chain?, IMHO all of us will be nearest to the recording and nearest to the live event: this is my target and this is what we try to do in our designs, nothing less: a very hard task btw but with lovely quality sound reproduction rewards for all of us.

We are starting the amplifier and tonearm design and that is our first target in those designs: we don't care if before or after our audio devices the deviations/distortions are bigger, as a fact this subject is what promote in José and I the Essential 3150 Phonolinepreamp self design.

Regards and enjoy the music.
Raul.

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.

"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!

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

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

Interesting reading, a bit heavy on methodology but the editorial notes and Dr Toole's response will give some background on what is actually possible and why we hear things the way we do.
This post started out as an attempt to educate ourselves to why and what are the things that matter to us humans and therefore must be attended to in design...

http://www.audioholics.com/techtips/roomacoustics/HumanHearingAmplitude.php

Hi Thom, having been in the business of balanced phono production longer than anyone else (since 1989- all the prior balanced phono circuit art were not *production* circuits) I can safely tell you that there is a lot of misconception brought out in the links you provided. I wish I had seen them when they were current!

The advantage of a balanced differential phono section is not that the cartridge will somehow act differently, it is the fact that the cable and the electronics act differently! To take advantage of the improved behaviors, we need to hook up the cartridge itself in balanced mode. This is easy as cartridges take quite naturally to this.

The result is lower noise throughout the phono system. In our case this allowed us to eliminate a stage of gain. That made the preamp more transparent, as it now makes less noise and distortion with wider bandwidth. IOW the signal path is actually simpler, not more complex, quite the opposite of the usual drone of balanced circuits being more complex!

Thanks ever so much for visiting this thread, Jonathan.

You make a very provocative point about varying sensitivity to different distortions (mechanical / speaker vs. electronic) which most definitely fits into the overall design process.

I have great respect for someone who has the courage to take on a complex design. It is a daunting task and one can easily get lost.

We need the electronic equivalent of the "Alpha Male" to push the collective envelope. Yes, some of them will suffer the electronic equivalent of the ill-fated Donner Party, but it's a choice they knowingly make, and we all benefit from the few successes along the way.

Cheers,
Thom @ Galibier

>>Again, your quality sound reproduction targets/priorities are a little different from ours<<

Indeed, and that is the beauty of freedom and liberty. As designers we can choose different paths. The result is that the marketplace is offered more choice.

I do not disagree with your approach. It is sort of what I used to do. However, chasing good numbers sort of limits potential. I am discovering that the only way to reach the next level is to get caught up in the emotion of the music. It is the unmeasurables that begin to matter. Great technical performance is a good start (I would almost say mandatory), but it can also be a trap. And that was the point I wanted to make. Don't get caught in a marketing game where specifications rule.

So in that sense, I am more in the camp of Thom. I can live with a +/-0.75dB RIAA error if the compromise gets me better connected with the musicians and their message. For me, this emotional connection is more important than absolute accuracy.

jh

Yes, the beauty of the marketplace is that we all have choices. I buck strongly when someone tells me that they have achieved sonic perfection and that I am a member of the great unwashed peasantry if I fail to appreciate their brilliance.

Sure, as manufacturers, we're all proud of our achievements. Why else would we make so many financial sacrifices to achieve what we consider to be lofty goals? Trust me - none of the small guys are buying 40 foot sailboats off of their audio income. This should be taken as a given (personal pride), and we as manufacturers shouldn't beat someone over the head with it.

I try to deal with my customers from a point of respect, realizing that there are many reasons for them to prefer another product - from personal taste as far as what a hi-fi system should do in order to bring them enjoyment, to the fact that they may want a piece of gear that is a bit more plug 'n play, to ... well, you get the idea.

So in that sense, I am more in the camp of Thom. I can live with a +/-0.75dB RIAA error if the compromise gets me better connected with the musicians and their message. For me, this emotional connection is more important than absolute accuracy.

I am reminded of an article that JC Morrison wrote in an old issue of Sound Practices. I wish I could find a copy of it on the web to link to. In that essay, Morrison broke the audio world into two camps:

1. The Audio Puritans - folks who would accept anything as long as they were told it was correct. Correctness frequently involved great suffering in the listening experience. No matter ... they are Puritans after all, and have a tradition to uphold.

2. Everyone else. Folks who actually want to have fun with their tunes.

Guess which side I gravitate to?

Cheers,
Thom @ Galibier

Dear Thom: Fortunatelly, like JH told us, there are different " roads " to meet the " border " and the freedom to do it.

Any one has its own " principles and priorities " and music bias in the sound reproduction.

Thom IMHO the best noise is no noise. Now, if we can't dissapear that noise then a good noise is " better " than a bad noise.

We are not " specs/number " lovers per se, but in some way we have to reflect what we are hearing through our designs.
We think that our designs not only have to " sounds great " but to measure great too and if we could achieve both targets that's will be great, don't you think?

I agree with JH about the emotional link with the music through and audio system, with out this interaction between the music and us it does not matters specs or design.
We try hard to have that link in the sound reproduction perception and at the same time to meet very low noise/distortions/colorations and accuracy ( like in the RIAA eq ).

Thom, I know that in theory almost all designers have ( between others ) these targets on mind but when we heard at those designs and when we read the specs I almost always ask me where/why some of them loose those targets. I'm not saying that we are perfects, far from that, but maybe we take care more in deep about simple " things " in the design philosophy, like José told us: the technology is there ready to help us, we only have to use it.

Jcarr: +++++ " 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. " +++++

I agree with this statement if the RIAA was a line instead than a curve where if one " note /discrete frequency " moves all the adjacents moves too usually more than 2 octaves, we perceive these kind of deviations like a colored sound: these colorations belongs to the audio device but not to the original recording.

I agree with Jcarr about: +++++ " The results of using NFB have a lot more to do with the capabilities and sensibiities of the designer than NFB per se ... " +++++

this is something that José and I discuss several times and the conclusion was the same that Jcarr posted: depend of the designer, where to use, how to use it, how much use it.

Btw, JH and Thom we are not marketing oriented ( we don't manipulate the signal to achieve a " marketing signature " or something ), we are on the audio device design because we like it and because almost all the designs out there can't achieve our goals/priorities in the music sound reproduction in the way we like it, as a fact we design to meet our targets not the consumer ones, we think that through our audio/music experiences, honest on the design, open mind and having the live event like reference we could meet ( or ve near ) the consumer priorities. Maybe we are wrong but it is the way we think.

There is no single parameter that define perse the audio device design, usually all designs have to meet several goals to be " listenable ", how different from others?: that's depends on the designers skills, designers goals and execution of the design in a finish product.

There is no perfect designer/design, we all have limitations of different kind and always have limitations on the quality/tolerances of the parts that we use it: all these parts have limitations and in theory those limitations will be our limitations. Other subject about is the cost/retail price that is a important limitation when we want to share/market the audio device between some price range level.

Regards and enjoy the music.
Raul.

Jose, or anyone else, have you correlated RIAA error to colorations? As in, error of x amount or of y type leads to a z coloration.

Dan, in my experience the term "coloration" involves 2 facets:

On one hand you have a measurable deviation of flat frequency response of X dB, sustained over Y Hertz of bandwidth, which will cause an identifiable sound similar to that produced by the bands of an equalizer. ABX tests have been conducted showing that the audibility threshold is lower as the bandwidth of the deviation is increased. This simply means that we are more sensible to this error when it spans several octaves. RIAA stages are particularly vulnerable to this kind of coloration, simply because the RIAA curve is made up of 3 turnover frequencies affecting big portions of the audible band.

On the other hand, coloration is also a characteristic sound caused by a circuit's more intrinsic factors that can't readily be measured with conventional frequency-domain analysis. Nevertheless it manifests itself as a "fingerprint" in the sound (punching bass for instance). This type of coloration, at times enjoyable, will reveal itself more with the passing of time. This is mainly the reason why ABX kind of tests commonly reveal the first type of coloration, but fail with the second.

Enjoyable or not, decreasing coloration is a good thing in order to preclude our ear from extrapolating the musical signal into a predictable sound.

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

This is controversial: if the audio device is " colored " and that colorations don't meet your music sound priorities maybe it does not like you even with your audiophile recordings. If those colorations ( faulty ones ) goes with your priorities you will love with all your recordings, this is fine with me but it is not for what I'm looking for.
Certainly I'm looking to enjoy my recordings, enjoy the music sound reproduction but never with " false " colorations/distortions, I can't tolerate this way because my ears/brain tell me that is wrong that it does not " sound " in that way in a live event ( of course that we are too far from the live event ).

The music perception is totally a subjective event and for this point of view every one have a different opinion and I respect all of those opinions. I'm trying to be nearest to the live event through be truer to the recording on our designs.

Some of us are more tolerant to the noise/distortions than others, I'm not ( and José is less tolerant than me ) and like you posted you are. Two different point of view from two different persons, nothing wrong with that these differences make our day fun.

Regards and enjoy the music.
Raul.

Dear Thom: This is the subject of your thread: +++++ " What Makes a Good RIAA or Line Stage? " +++++

We all already talk about many subjects for a better audio device design and I think that other one is lay-out, two same designs could sound different with different lay-out. Not many people think seriously about but the circuit board layout is of paramount importance for the performance in audio devices.

Regards and enjoy the music.
Raul.

Hi Raul,

Yes, layout is indeed important. Jim Hagerman is too modest to admit it in this thread, but he put enormous amounts of work into the layout of the Trumpet. I recall him mentioning that he went through some 14 major architectural layout changes.

In order to get the best performance out of a single chassis design, he kept returning to the two tiered layout. Yes, grounding schemes and general layout plays a big part the final product - especially one so sensitive to noise as an RIAA stage.

We're all coming from a perspective that all of the components under consideration are at a very high level of resolution - that we're all after a design that will give us as much resolution we can get. I'm debating those special components which make it past the final cut if you will, and "fun factor" has to be a major acceptance criterion. Anyone who tells you otherwise is either lying or is a tortured soul.

Surely, once you've peeled away layers of distortion, input overload and such, you can never tolerate these flaws in a component. As I've written on several occasions, a poorly designed RIAA stage can overload and sound uncannily like tracing distortion. Nobody wants this, and yet there is a surprisingly large number of highly regarded components which exhibit this and other design flaws.

-----

This thread is taking a philosophical turn, but I think that's o.k. because it gives us insight into the whole person - be it the designer or the end user.

I always look at any design as a manifestation of the designer's personality. I've known a couple of audio designers who could be characterized as having obsessive compulsive disorder. Their designs were equally dysfunctional. In spite of their painstaking attention to detail, their end product was as flawed as their personalities were.

I am not arguing for being sloppy and careless in stating the above. Please do not misinterpret this as being the case.

I have to "out" you on this whole "false colorations" thing however. Your "it has to be perfect" mantra is really tiring me out. You are beginning to sound like one of those Audio Puritans.

Given a choice of a "correct" design (whatever the heck that is) which doesn't allow me to enjoy 30% of my record collection and a "flawed", colored design which allows me to pull out ANY record in my shelf without having to ask if its sonics are "worth" putting on my record player, I'll take the latter in a heartbeat.

I can't begin to count the number of components I've owned which took all of the fun out of hi-fi because they were "accurate".

I'm sorry, Raul but as good as the Essential is, it is as colored as many of the other fine RIAA/line stages I rank in the top tier (and the Essential is a fine piece). Please get over it and realize that no one can be all things to all people.

Is the very fine Essential is more harmonically correct than many of the other fabulous components out there? Absolutely not. The Essential to my ears is very much an Avery Fischer Hall type of component - a very lean and modern sound. Other fine units we've discussed in various threads cover other parts of the sonic spectrum - ranging towards pre-renovation Carnegie hall, for example.

Which one is correct? Both are correct and neither is. At some point, one is forced to choose, and here's where the "fun factor" helps to break the tie.

I'm coming down on you hard, because in your writing, you are portraying yourself as being 180 degrees apart from my sense of you and your goals after our fun day together two weeks ago. I think you are more in the "audio fun" camp than you prefer to admit in public. This may be a language thing, but consider yourself "outed". Please don't redouble your efforts to prove me wrong. Resistance is futile.

Close your eyes, take a deep breath, and let your hair down. You'll feel better. I promise you.

Regarding Audio Puritans, I will go so far as to discourage perspective Galibier customers from buying one of my turntables if they give me so much as a hint that they are Audio Puritans. Life is too short to work with someone whose sonic goals are that different from mine. I would prefer that they purchase a mainstream turntable and let me spend a weekend climbing a cliff or going skiing.

-----

About marketing ...

"The lady doth protest too much, methinks"

You don't need to tell me that your work came out of passion, because as I mentioned earlier, none of us are buying 40 foot sailboats from our income in audio. All of us are about passion, and I applaud yours and Jose's efforts, because you have achieved something very special. Is it better than everything else out there? Absolutely not. Let's not create a mythology here. This is what I object to.

We need to return to the subject of this thread. I'm getting tired of this.

Cheers,
Thom @ Galibier

Raul, give it a rest. You're making the same points over and over again.

The salesmanship is getting old.

Thom, a replay curve that deviates more from the RIAA standard can sound better on select recordings, but it will almost certainly also sound worse on other recordings, compared to a replay curve that deviates less. My experience is that a flat RIAA curve is likely to allow you to enjoy more of your LP collection, not less. If 0.1dB or better is possible, go for it, as I think that on the whole, you will be ahead. The one categorical exception is when an LP contains sampling noise (sometimes CRT monitor noise). There are albums by Kraftwerk and Lorie Anderson that have this which I find painful. Everything sounds find until the sampler kicks in, and then I look for a wad of cotton, or wish that I'd designed an RIAA playback network that shelved down the top end (grin).

As an aside, the other possible solution for an RIAA playback curve would be to implement an EQ trim control, like with the FM Acoustics designs, or maybe a Cello/Viola Pallette. I've listened to and played with both, and yes, I can see their point.

I also find that a theoretically "better" solution - better power supply, better regulators, better amplification circuitry etc. will nearly always improve the sound of nearly every LP that you own. I don't find that "more accurate" means that you become more picky about the LPs that you can find enjoyable. Yes, you may become more aware of recording, EQ or mastering issues, but the music and performance comes through even more strongly, more than enough to overwhelm trivial concerns about the recording. The better my designs become, the more I appreciate a greater number of musicians.

I do find, however, that when it comes to component selection, you have to use your ears and subjective taste, in addition to your head. I've picked components that on paper should have been the cat's meow, but in listening turned out to be a pig's kiss instead. The designer cannot know each minute particular of every component that he chooses, and as they say, the devil is often in the details. So unlike the case with overall topology or circuitry or layout, with components I find it necessary to have a "range of candidates" on hand and go with whatever sounds the best - in the context of the circuit being tested. Engaging in this is more like cooking or choosing clothes than it is intellectual design, and is the phase where the more artistic types can strut their stuff, and pull level with or even ahead of other designers who may be their intellectual superiors. That's the fun part about audio (designing it as well as using it) - there is a place for the sensibilities as well as the intellect.

regards, jonathan carr

Dear Thom: With this I'm finish:

+++++ " Your "it has to be perfect" mantra is really tiring me out " +++++, that's what I posted: we are different in this subject but I can asure you that we have more in common than differences about sound/music reproduction.

+++++ " I'm sorry, Raul but as good as the Essential is, it is as colored as many of the other fine RIAA/line stages I rank in the top tier (and the Essential is a fine piece). " +++++

Absolutely, I never speak that it is not only that we try to leave those distortions/bad colorations at minimum, that's all.

+++++ " Please get over it and realize that no one can be all things to all people " +++++

Absolutely, I don't think in other way: I agree.

+++++ " Regarding Audio Puritans, " +++++

No. I'm not: I'm only trying to be better.

+++++ " a very lean and modern sound. " +++++

No, it is not ( I don't know what means " modern sound " ): the sound perception depends on the whole system and my point of view about is that our design is very high revealing of what happen in all the audio chain because that design does not hide almost anything out there, of course that the Essential is part of that audio chain and puts its " grain of salt " about.

The diffrences is how you, other people and I perceive the sound reproduction in an audio system and how be related against the live event or at least near to the recording.

Our attitude/philosophy in the electronic audio design is: truer to the recording and you can't blaim me for that, it is only a way to think that is a little different from yours and other people but there are other people that think in the same way that us: our design goes for that people and we will wait that the other people could change ( a little ) in the future, no our design is not for all in the same manner that horns speakers are not for every one.

Thom, thanks to our differences we have some fun and we can learn a lot about those differences, don't you think?

Regards and enjoy the music.
Raul.

Raul:

Our attitude/philosophy in the electronic audio design is: truer to the recording and you can't blaim me for that

You probably mean "truer to the reference riaa de-emphasis curve". Whihc, hopefully, means closer to the recording. After all, the phono equaliser only equalises whatever is picked up by the TT setup and transfered through the wire & connectors...:)

BTW, Raul, do you have an added pole at 3,18us as well as 75us?

Dear Greg: You are right.

yes, we added the 3.18us pole. You can choose it with an internal " switch ".

Regards and enjoy the music.
Raul.

Thanks, Jonathan.

You've filled in the blanks better than I could hope to do, so I won't elaborate (too much). I completely agree with you that there is a difference between hearing flaws in a recording and being annoyed by them. A very good friend of mine runs a Lyra Olympos cartridge which exhibits exactly these positive attributes.

Perhaps a good mechanical analogy to what we're discussing lies in tonearms. A world-class tonearm can allow the turntable and cartridge to better do its thing by more effectively dissipating resonances. In being less resonant (perhaps a better term is "appropriately or artfully resonant"?), you can hear more of the music in the groves while at the same time ticks and pops fade to black much more quickly and become less annoying ... more of the good stuff and less of the bad.

The conversation about RIAA tolerances as well as the frequency width that you and Jose have been exploring is a provocative one. I think that all of us agree (at least no one has corrected me on this point yet), that getting low RIAA deviations is not so much a design constraint as it is one of labor and parts cost (component matching) to achieve the correct turnover points.

The meaningful threshold beyond which RIAA deviation becomes "specs-manship" and nothing more is what Jim and I were calling into question. I can't say for certain where this response threshold is, but your comments about the ear's higher sensitivity to electronic colorations over mechanical ones (speaker, room interaction, etc.) makes much sense, but still doesn't tell me whether .05% is just enough or if it is excessive overkill.

On a related issue, I'm curious about any psychoacoustic effects as far as the width of a frequency response deviation is concerned. I suspect we have several different thresholds along a continuum.

What I mean is that a very small width deviation might go unnoticed, whereas a slightly wider deviation (but still a spike) might be perceived as an anomaly, and a slightly wider band deviation may well be masked, or blended in. As Jose commented, these wider band colorations may well take some long term listening in order to be recognized and become potentially bothersome.

Some of the Lamm electronics exhibit such a wide band coloration which is quickly recognizable in completely different system contexts. While the gear is highly resolving, the colorations are very noticeable. I would never criticize someone for loving a Lamm, BTW, otherwise I'd have to own up to being an Audio Puritan (grin).

I have no insight into psychoacoustic experiments on this subject, and if someone does, I'd be interested in learning about it.

As Jim Hagerman (I think it was Jim) commented, we have to start from a technically correct baseline if we have any hope of coming up with a design that inspires us. I think everyone is in agreement that excellent technical performance is a necessary but NOT SUFFICIENT condition for greatness.

The good news (for me) is that when I get lost in a design change, my astute, musically trained wife drops by and immediately tells me whether we have music or merely hi-fi. Many of us are lucky to have perhaps the finest measuring tool known to man ... a smart, sensible wife who understands the goal of hi-fi.

And yes Raul, Jose's brilliant effort is one of those very fine, top-tier components which achieves greatness. I have difficulty simplifying my English to help you with many of the subtleties in my writing.

I try to be clear ...

It made perfect sense to compare component colorations to those of concert halls. The language has been in our hi-fi vocabulary for some 30 years. I look at the term modern in a component to be analogous to the frequency bias of many modern concert halls like Avery Fischer. As an aside, I've heard that Portland has a fairly new concert hall that resembles many of the fine halls of the past with more of a mid-bass and lower midrange center of gravity.

Unquestionably, the Essential is Avery Fischer Hall, and I've received quite a few private e-mails to back me up on this. Is Avery Fischer a bad hall? Absolutely not. Do some people prefer the hall in Rochester, NY (can't remember its name). Certainly.

Back to our regularly scheduled programming ...

Cheers,
Thom @ Galibier

Dear Thom: IMHO thanks to our quest of exellence in the audio performance this thread exist. That quest on exellence bring us here.

I know exactly what you like to hear unfortunately you don't know the same about me.

You are a very well respected TT designer and I wonder what is your quest through your great TT design?

IMHO, I think that we have to look for " evolution " and this is what we are trying to do through our designs.
We always think that there are " out there " a better future for all of us and a better way to make " things ", we are on the quest of it.

I know that we have " to fight " not only against limitations in electronic parts, technology limitations but more important than that limitations in the way people think: this is our challenge, 90% or more of the Essential 3150 ( presentations ) were on tube lover audio systems, not an easy task I can tell you.

We try to be " open mind ", many people ask me why don't tubes?, my answer ( till today ) was always the same: we use all kind of technology that could help us to achieve in the best/better way our sound music reproduction targets and unfortunately the tubes can't help us, not yet: maybe in the future?, maybe: who knows?

Btw, my " hat off " to all non comercial mind tube designers because it is a great really great challenge to achieve " decent goals " in the Phonolinepreamp design with that technology: congratulations!!°

Thom, I know that you are the " boss " in this thread but I'm only trying to help to all of us with a little different point of view, at least from yours.

Regards and enjoy the music.
Raul.

Hi Thom, I thought I ought to point somethings out. A phono section needs to be really accurate. Many seem that way, but are not as they add something in trying to be 'detailed'.

I have found that high overload is a good thing. So our MC section overloads at about 250mV so that even a high output moving magnet can't overload it; at overload the output is making over 120V peak to peak.

What happens is that less than perfect recordings can thus still be enjoyed. IOW bad sounding LPs should be utterly playable on the best of systems without a lot of fatigue while the best sounding LPs transport you to the music.

In practice this works and yet our RIAA accuracy is within 0.05%. We extended the curve all the way to 100KHz and the low frequency cuts off at about 1.5Hz, so tone arm resonance defines the lower cutoff. In order to pull this off the power supply has to be very very stiff, and I think I mentioned before that we created a proprietary regulation setup just for this purpose (our regulators are quieter than the 'Super Reg' for example).

Differential amplification comes into play here as differential amplifiers have a 'cross mode rejection ratio' which is an ability to reject noise in the power supply. This is further enhanced by using two-stage constant current sources that are also designed to reject power supply noise.

Overall this gets you is a phono section that is unperturbed by poor recordings. It does seem if a phono section has issues, that often bad recordings will reveal that more than good ones!

"It made perfect sense to compare component colorations to those of concert halls. The language has been in our hi-fi vocabulary for some 30 years. I look at the term modern in a component to be analogous to the frequency bias of many modern concert halls like Avery Fischer. As an aside, I've heard that Portland has a fairly new concert hall that resembles many of the fine halls of the past with more of a mid-bass and lower midrange center of gravity.
Unquestionably, the Essential is Avery Fischer Hall, and I've received quite a few private e-mails to back me up on this. Is Avery Fischer a bad hall? Absolutely not. Do some people prefer the hall in Rochester, NY (can't remember its name). Certainly."
Thom: with all due respèct but your analogy of comparing a concert hall to a phonostage does not make sense to me at all. Certainly as a profesional muscian myself I can tell you that Avery Fisher hall is one of the worst concert halls ever build. The NY philarmonic has been trying to get away from there for years (concerts at Carnegie) although they are stocked there due to the Fisher foundation legal issues. Coming back to Raul,s phonostage imho it will sound as Avery Fisher hall if thats what its feeded but it will also sound as Amsterdams Concertgebouw,or Boston,s Symph. hall(great halls)if thats what is played through it.
The problem with comparing a phonostage with a concert hall is that lets say for ex: The Musikverain in Vienna has sort of a rich full bodied warmth that the Vienna Phil. players have come to adopt as part of their sound and playing tradition; that it in itself some may say that its a colored sound but again its their sound in an aesthetic and artistic conception. The problem comes when you try to convery those same parameters to an electronic component and what comes out of it is going to be a an edited deviation of a VALID artistic coloration.
As for myslef , i,ve rather try no to edit and stay as true to the source and not only in the phonostage where I think counts not as much as with speakers-room.

Thom:

Regarding playback eq deviations, a small width one may indeed be rarely noticed. Wider band deviations will almost certainly be noticeable if you have a more accurate playback curve at hand for comparison (or have experienced one recently), but if said wider band deviation is the best that you have experienced (or some time has passed since you listened to a more accurate network), maybe you won't mind (or notice). However, although I haven't measured the Lamms, I know from my own work that what we perceive as measureable frequency deviations (as would be the case with an improperly designed RIAA network) may not always be so. Component choices, HF bleedthrough via the power supplies, resonances in the RF range all play a role in the perceived frequency balance. For example, although it appears to be accepted knowledge now that different capacitors (or resistors) have their characteristic signatures, the same also applies for active devices (even if they conform to the same nominal spec). If I don't like the perceived frequency balance that I am getting, it is therefore not a problem to change that while keeping the measured frequency response in the audible band the same. The process may involve some trial and error, and it may take me a few tries to get where I want, but it certainly can be done.

Regarding the analogy with concert halls, I get your point, but I am not sure if it is on target. Many instruments have very different tonal balances depending on the angle and distance that you listen to them from, and you need to physically put your ears where the microphones are to verify whether what you think you should be getting is really what is inscribed on the LP or not (and don't forget that mikes have different frequency responses from our ears). I am fortunate enough to have friends who are recording engineers and have allowed me to sit by the microphones (sometimes on a ladder!), tap into the mike feed, go back to a normal seat in the audience, listen to the analog tape master on the same day, and then a few days later, listen to the lacquer masters. Very, very educational. I encourage you to search out opportunities to experience this.

I very much agree with Ralph's comments on the desireability for high overload margin, and I will add that this is needed at ultrasonic frequencies as well as audible ones. Groove dirt and damage played through the cartridge manifest themselves as transient impulses (very high amplitude, very high frequency content) that at least the front end of the phono stage needs to deal with. If the phono stage doesn't have good overload margins and recovery, pops and ticks will be emphasized, so will record noise in general, and this can also shift the perceived tonal balance upwards so everything sounds brighter than it should.

To add another point, good behaviour in the RF region is also desireable, because there is enough energy (particularly in the 500kHz~ 2MHz range) normally reaching the phono stage that, if the phono stage has problems in this range, IMD can result in inharmonic distortions subheterodyned down into the audible range. Obviously, AM radio stations broadcast in this band, and need to be dealt with. However, phono cartridge loading can also generate resonances in this same region. The inductance of the cartridge's signal coils will react with the capacitance of the interconnet cable to create a resonance in the RF range. Let's take a Denon DL-103. Measuring, I get 40.5uH coil inductance. phono cable capacitance 150pF, resonant frequency 1.94MHz. Now let's see what happens to the measured frequency response when we vary the input load resistance of the phono stage. With a load of 47kohm, the electrical response is flat out to 100kHz but starts to rise, and by 1.94Mhz it is about 7dB up. If we say that the correct load resistance is sq.rt. (L divided by C), we get 500 ohms, and while the frequency measurement looks the same as with the 47kohm load, it stays more or less flat out to a -3dB point of 1.77Mhz. Even if we load at 270ohms, approximately half of the optimal 500ohms, the frequency response still stays flat out to 100kHz, and at 1MHz, we are only down by 2dB.

So, even when you give a low-medium input impedance MC various loads, the audible frequencies are not directly affected. The measureable frequency variations are occuring at ultrasonic frequencies. So why do people report major difference in sound when the input loading is altered? IME, HF behaviour of the phono stage and IMD is the answer. IOW, if the phono stage has exemplary behaviour at RF frequencies, whether the triggering source is a radio station or a resonance between the coil inductance and cable capacitance, that stuff will remain at RF frequencies and you won't hear it (at least not easily - grin). But if a sensitive part of the phono stage has performance issues at those same RF frequencies, IMD will make it far more likely that, for example, changes to cartridge loading result in big changes to the sound. And listening while altering the input loading of phono stages with high HF overload and good RF behaviour as compared to those that do not, bears this out (at least in my experience).

Do note, however, that since coil inductance and cable capacitance determine the resonant frequency, with enough coil inductance and capable capacitance, the resonant frequency can drop to within or close to the audible range, and the likelihood of hearing the effects becomes far higher, regardless of how well the phono stage may do at RF frequencies.

If the designer has taken this sort of stuff into account as well as obvious things like an accurate RIAA network and low noise, the greater the chances are that all of your LP collection (or at least more of it - grin) will sound good.

Again, I agree with Ralph that nasty recordings are often a better guide to the real worth of a phono stage than kind ones. Usually, when I am testing or auditioning equipment, I prefer to put on "system-breakers" - recordings that I know from experience have a good chance of throwing a system into fits. None of that sissy audiophile stuff! (^o^).

regards, jonathan carr

Hi Raul,

If I may offer you a bit of friendly advice, and remember ... free advice is usually worth a bit less than what you pay for it.

I know that we have " to fight " not only against limitations in electronic parts, technology limitations but more important than that limitations in the way people think: this is our challenge, 90% or more of the Essential 3150 ( presentations ) were on tube lover audio systems, not an easy task I can tell you.

This is where you need to be patient. You will not convert everyone, and most of those whom you do convert you will not do so overnight.

I think I can state with confidence that each and every one of the designers participating in this thread have the same amount of pride in their product that you do, as well as the vision that they have a unique window into musical reality. I would expect no less.

Fighting the limitations in peoples' thought is one of those Zen paradoxes however. The more you try, the further behind you get.

It's important to take a historical perspective on this - to realize that many great concepts did not benefit the innovator ... until years after their death. Now, none of us are arguing that we like this, and many of us have achieved some degree of notoriety in our lifetimes (still waiting on that 40 foot sailboat), but one still needs to accept the possibility that success (no matter how you define it) may not be in the cards for you.

There are all sorts of reasons why consciousness moves slowly. Certainly, people are slow to move out of comfort zones. Have you ever heard the expression: "whom are you going to believe? Me, or your lying eyes?". You have to accept that people change at their own pace, and you can't force your reality on others. If you push, they will push back.

Oh yes ... the last thing I want to do is to be the "boss" of any thread. I am humbled by the great minds who are participating here.

Cheers,
Thom @ Galibier