What Makes a Good RIAA or Line Stage?

Dear Thom: Ok, its done.

Ralph bring here a critical point in the phono stage: high overload, and not only is a good thing but a necessity.

We found that with increment on the overload we can have lower distortion and better quality performance all over the frequency range.

" that often bad recordings will reveal that more than good ones! ", IMHO both could tell us a lot of the performance designs: in a good designs the bad ones will sound " less bad " and the good ones a lot better, the average sound will be " good sound ". In a " bad dsign " the bad recordings will sound unlistenable and the good recordings only " ok ". At least that's what we experienced about, others could have different experiences.

A good design is the sum/add-up of many subjects and the right synergy between them: an accurate RIAA eq per se means nothing if it not coming along: low distortion, low noise, high gain, high overload, high common mode rejection, right lay out, precise ground planes/star grounding, wide bandwidth, low output impedance, right output attenuator/volume control ( by the way this subject is the Aquiles heel in many designs. ), linearity, execution/build design, etc, etc. and many other parameters that you already posted about.

The challenge is to link all those parameters when some of them " fight " one against " other " or when we have to fight with non-linearities or high order harmonics like in the bipolar design that we are using.
Here it is when we have to use not only the technology that we can reach but the experience, know-how and skills that every single designer has and that is different form each other, that's why exist several differences/approach on the Phonolinepreamp designs, some ones better than others or simple differents.

Obviously that any single of us could think that our design is the best one and many of us could think that we can prove it. We think that our design is different.

Regards and enjoy the music.
Raul.

Hi Ralph, Jonathan, Raul ...

Back to our regularly scheduled programming. This is good stuff ! I need to take it in small chunks.

This concept of immunity from input overload is a critical one. Listen up to these wise folks folks.

If you're scratching your head and wondering if you screwed up your tonearm setup (or if your cartridge is unable to track a passage), try borrowing another RIAA stage ... hopefully one which is known to be both immune to input overload as well has having a lightning fast slew rate (the ability to respond quickly to transients without overshoot).

If you're not practiced at listening for this, you may well assume you have a mechanical issue to resolve (cartridge or tonearm setup) when the real problem may be with your electronics. Don't beat yourselves up about your setup skills until you validate what you're really hearing.

Cheers,
Thom @ Galibier

A quick note on riaa fm practical experience only. The necessities: Overload margin, stabilised power, fully corrected and stabilised curve, channel identicity. The last two are more than just difficult -- they're horrendously painstaking, boring (think of "trimming" to get the "right" R -- and once you get there, you realise that your next pole is off...), and expensive: anyone ever try to really "match" components? -- and I don't mean those that do this for a living... Goodness! Trying to match the 1st stage loading before riaa was a trial... let alone further up.

This is only fm a small experience trying to make my own phono as, the ones I could afford were miserable and the ones I wanted were out of reach financially.:)

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

This statement/experience is of paramount importance to understand the statement " truer to the recording ".

A week ago ( during my visit to San Diego ) Norm, Tim and I were talking exactly what you posted: that what we have to " hear " is what the micro's " take " and not want to hear what we perceive at our seat in the hall: that's too different, a lot different.
This is an e-mail that I send to those people following on that subject:

+++++ " De: Raúl Iruegas [mailto:silviajulieta@prodigy.net.mx]
Enviado el: Martes, 20 de Febrero de 2007 12:10 a.m.
Para: 'ctm_cra@yahoo.com'
CC: 'timryanbmw@yahoo.com'
Asunto: Essential 3150.
Importancia: Alta

Dear Norm: Our “ hot “ conversation last night was really “ teaching “ for all of us ( I think ).

I would like to share with you some additional thoughts about:

For many years I asked why the sound of my system was more transparent, better soundstage, tonal balance, etc, etc than the sound that I heard at the music hall, after some time suddenly I “ find “ that the recording microphones have a very different “ seat “ position than mine, it is not only that the microphones are nearest to the source ( instruments and instrument “ room “ area ) but in more “ clean “ environment: Tim, you and me normally seated at 6-8 rows ( 20-25 meter ) where the sound that we perceived against the one that goes through the microphones are truly different ( not only by the distance subject ) for the dispersion ( reflections, diffusion, hall time delay, etc, etc, ) through the absorbing/diffusion of the “ hall environment that includes all the people that surround our seat place.

Now, the sound engineer on the recording process always try to choose the best microphone for its quality reproduction that include ( between other subjects ) very wide frequency response ( this depends on what kind of music they would to recording ) and the lesser frequency range deviation from flat: this means ACCURACY. With out this accuracy the sound recording will be more colored and normally the sound engineer does not want to have a “ colored “ sound session. This same sound engineer choose the best monitor speaker he can get and the main subject a bout is a “ whole “ accurate characteristic monitor speaker, not a colored one ( I hope ).

So, the recording must be an accurate one to the live event.

What is the “ critical mission “ that any ( decent ) phono cartridge has? ( TT/tonearm ), well to try to reproduce what is on the recording in the better accurate manner, with the less distortion/noise/colorations.
This signal that comes from the phono cartridge has to pass through the Phonolinepreamp and this audio device has the “ absolute mission “ to reproduce in an accurate way ( yes I know that other subjects about, but please forgot for the moment ) exactly what the cartridge get from the recording that was made through the microphones that were in a different position that our hall seat that is the way that you like to hear in your home.

Norm, I want that you think in deep about not thinking who is right here but using common sense and trying to help the true music reproduction. It is not a fact that what you like or what Tim and I like but what is the “ true “ or near the truer.

This “ truer to the recording “ accuracy approach does not means : analytical, cold, polite, lean, etc, etc sound reproduction, that I´m sure the Essential 3150 does not have, it must be accurate and “ pleasant “ and this is what we have to look when any one heard the Essential 3150. I´m not saying that the Essential 3150 already achieve that goal in a perfect manner but it is very near to that target.

In my humble opinion all the music lovers must support that kind of audio devices design approach because in that manner all of us could help to the music and most important could help to growing up faster the quality reproduction on the high-end audio device performers: any.

Anyway, we really had a great time during my visit.

Kind regards.
Raul. " +++++

Regards and enjoy the music.
Raul.

Raul, this is a different thread. I have just read your same post in your Essential thread. Why do we need two Essential threads. Please give us a break.
Thom attempted to start a new thread exploring some very interesting ideas regarding phono stages in general. Several designers have responded and have made this a great thread for some of us to learn new things about design philosophy. Please, please give it a rest.

Your either "Type A" listener or "Type B" listener.

- W

...fully corrected and stabilised curve, channel identicity. The last two are more than just difficult -- they're horrendously painstaking, boring (think of "trimming" to get the "right" R -- and once you get there, you realise that your next pole is off...), and expensive: anyone ever try to really "match" components?

Greg, this is exactly the same reason that caused Dr. Stanley Lipshitz to express this words 28 years ago: "To begin with, trimming is a difficult procedure, for each component affects at least two of the finally realized time constants of the network. Furthermore, to be able to trim accurately one must have either a precision RIAA circuit for reference or else be able to measure over a dynamic range of >40 dB and over a frequency range of >3 decades to an accuracy of tenths of a decibel. This is not an easy task".

Fortunately enough, nowadays we have DSP technology, which can now be used to address precisely this task. Part of my research in the last few years has been to create an effective trimming procedure that allowed me to calibrate the RIAA with a resolution of thousands of decibels (not kidding). I believe having an accurate RIAA is audibly superior, for the same reasons that Mr. Carr mentioned, as well as many engineers and enthusiasts have investigated.

WOW! A great thread with many designers I respect. The seam I've been mining all my audio DIY life is phono stages.

There is so much reward for the effort and of course, a little frustration on the way :-)

As most here will know, a lot of analog's perceived "issues" are actually present in the phono stage and quite often attributed to the mechanical aspects as Thom alludes to.

The overload aspects of the phono stage encompasses all aspects of the design and is where many designers, not so here though, look at the basic requirements and assume the signal levels are low and limited to the audio stage
only. Compared to power amps they are but I think many look at headline specs, say a MM input of 5mV and leave it at
that. However take that figure at 1k and then project that to 20k which is 20dB higher and then things look way different. Include the ability to handle HF transients and things look different again. I only know any of this
through trying stuff out and listening and it is illuminating to have people put a technical perspective on this i.e. some of this stuff can be measured.

However the thing I design for is hard to measure OR may be measured if only we knew what to look at. I design for
the ability of the music to communicate it's message to me. Consequently I don't measure anything. Mainly because I can't as I don't have the equipment nor could I afford it of sufficient accuracy AFAIK. Knowing (I think) the
accuracy required to measure RIAA EQ deviation, I find it hard to believe some of the figures being claimed in this
thread. A deviation of 0.1dB is I think 1% accurate. To measure this accurately requires test instrumentation to be
an order of magnitude more accurate so that's 0.1%. Is test equipment of this caliber being used to verify this? And
regularly checked against a test standard? As an amateur, I go for the "model it" approach aiming for the best fit
to the curve and then used better than 1% parts to hopefully get within the 1% window so within the 0.1dB deviation which could be called +/- 0.05dB. Is it? I have no idea but as I have got the model better and my design has got better, it would seem so.

Talking abou the EQ curve, there is the matter of the cutterhead rolloff or the 3.18uS turnover point. I used to
approach this as a lone point however as someone pointed out to me once, this turnover point affects the curve
waaaay lower down and so now I sim it as a LP filter that my inverse EQ feeds the sim with. Correcting for it's
effects lower down got me closer to a better phono stage. So I would question in some way, the effects of making it
switchable IF the other components in the EQ are not also altered at the same time.

I found it real interesting Johnothan's calcs re the effect of cart resonance and the frequencies it occurs at. It got
me thinking about its effects on not only the audio cct but the power supply. I personally don't like regulation (we
all have our prejudices) as I like to think the power supply should be as Nic Doshi says, as benign as possible. My experience is that this is really hard and when using regulation, for me at least, it has proven to be impossible.

With all this HF stuff going on, the concept that most regulators have a vice like grip on proceedings is a myth and their contribution to the circuit becomes nearly as great as the audio circuit itself. I try and avoid things like that ... well in my head anyway :-)

It will be no surpirse therefore that I favour hollow state. Another area that seems to be a poor cousin in the
power supply design is the heaters. IME, they have as much inluence over the sonics as the HT and so in mine, they
get as much filtering as the HT. It's not surprising that many commercial offerings kinda pass this by as it becomes
real expensive to do. I have no idea what anyone here does so it's not a criticism, just an observation of those
circuits I've seen.

I find it at odds with my experience that you can have a great design that makes good records sound good and also
bad resordings sound good. IME, a great design makes all records sound better and that's not done by introducing
"flavour" by being coloured but by making it more technically able. Bad recordings, to me anyhow, usually present more of a challenge and a better design is able to meet this challenge without causing the circuit to hold a white flag up and sound horrible. As the design gets better, good recordings sound even better, bad recordings can also benefit and rise above awful to enjoyable BUT the real essence of a good design is that all recordings sound more different. It is this aspect that drives me on. The better the design, the more of my record collection opens up to become enjoyed.

Some that used to be shocking I have discovered are real treats now. This is why I focus on the communication of the
music being the sole arbiter of goodness in the design and as such, the rest seems to come with it.

Another thing I wonder about is the concept of channel seperation. A cartridge is only so so in this regard. Once I used to do dual mono re the PSU but now I use a single supply as this seems to ground the musicians better and they seem to play together better. Its seems to be one of those HiFi vs music trade offs. I also wonder how you can get two supplies to be perfectly the same regarding noise and grounding and so therefore be exactly the same at all frequencies. As I can't see how to do this and the results of a single supply in my designs to be superiour, I wonder how much we should chase this notion of chqannel seperation.

Thanks Thom for kicking this thread off. I hope it reveals a bit more as it goes. I just wish I came to find it earlier.

regards,

Stephen
www.izzy-wizzy.com/audio

Hello all, Is my assumption correct that a small deviation in the frequency response at the phono stage is magnified by the time it reaches the speakers? In otherwords if you have a 1db variation at the phono section it is much more detrimental than a 1db variation at the amp or the speakers?
Bob

Dr. Stanley Lipshitz also published a set of simple formulas for exacting RIAA reproduction. His articles on the RIAA curve can be regarded as reference material.

I can vouch for a number of StephenR's discoveries. He mentioned the sonic importance of heater supplies, and how they affect the sound quality just as much as the B+ supply. It may sound crazy, but he's right. In my linestage, I have tried different heater supply components and they affect the overall sound quality just like swapping components in the B+. Filter caps with the same value capacitance can sound quite different. The biggest surprise for me was trying a large choke filter in the heater supply. I used a .15H choke with a DCR of only 1 ohm, which is larger than most power amp transformers. The sound of the linestage is much improved with the choke in the heaters. Go figure. There's just no substitute for keeping an open mind and trying different circuits and different components to find the one that is "just right."

Dave

Dave

>>is frequency response at the phono stage is magnified by the time it reaches the speakers?<<

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

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

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

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

www.hagtech.com/images/accuracy.gif

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

He got it right. Resonances can be ultrasonic.

www.hagtech.com/loading.html

More info on RIAA at:

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

jh

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

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

Regards and enjoy the music.
Raul.

>>3.18uS turnover point ... switchable ... other components altered at the same time<<

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

jh

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

jh

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

regards, jonathan carr

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

Quite easily in fact (with the right topology). The 3.18 us turnover point is an extension to the RIAA de-emphasis curve, whose purpose is to compensate for the practical considerations that the RIAA neglected when the standard was created. In this case, that the cutting heads at the mastering facilities were unable to pre-emphasize up to infinity, but instead they were only garanteed up to a certain frequency. The Neumann heads used this turnover point, and they have always been very popular. Many LP masters were cut with this heads.

The 3.18 us turnover causes a phono stage to stop its rolling-off of the treble at a frequency of roughly 50 kHz, which in turn causes an added sensation of "air" in the reproduction. However, this tool should be used carefully because some MC cartridges generate too much ultrasonic energy (ringing) when excited by the cliks and pops in the record.

On a scope display, the ringing looks like decaying high-frequency tones superimposed on the audio signal. Although this tone is beyond hearing, it could potentially cause trouble to the amplifier or speakers.

In my opinion, the solution is to have the option available with a switch, and leave the choice to the listener. This provided that the manufacturer actually understands that the 3.18 us is a zero in the transfer function, not a pole as some designs Raul and I have seen!

In my opinion, the solution is to have the option available with a switch, and leave the choice to the listener. This provided that the manufacturer actually understands that the 3.18 us is a zero in the transfer function, not a pole as some designs Raul and I have seen!

By this do you mean, there is the RIAA EQ curve and the transfer function is a mathematical way of expressing it. The 50kHz cutter head rolloff is not actually part of that but practically is a LP filter applied after the RIAA EQ in the recording process.

regards,

Stephen

Stephen, exactly. Having the 3.18 us zero in the phono stage simply leaves the response flat after 50 kHz (instead of keeping the roll-off up to infinity). That way it cancels the 50 kHz filter that was not part of the RIAA eq. With all things being equal, the sound would theoretically be closer to that of the master tape (or the live event in case of a direct-to-disc recording).

This thread made me re-visit the interface between the cart and the phono preamp. My thanks to Jonathan Carr for his insight on this. It has taken my preamp onto a new level.

cheers,

Stephen

Daer Thom,

After speaking with you yesterday I revisited this thread as you suggested in reference to the cartridge being or not being a balanced component. I am convinced that there is no downside to treating the source pickup as integrely balanced. I think the wiring of the tonearm from cartridge clips straight through to the phono pre line stage is the best way to exploit what potentialy can be a sound advantage in a fully balanced system.

I think that the advantages of a fully differental balanced system with first rate Phono Stage design intergrated into the Line stage to avoid signal degredation through interconnects in concert with seperate stand alone dedicated power supplies says a lot about what makes a good phono pre line amplifier.

I am so glad I did take a second look The first time I was here it left my head swimming trying to understand the logic of the technical positions presented here. I have learned no small amount since then, I understand more yet I still am content to just soak it up and keep listening to all the points of view.

Some things may feel repetitious at times but my experience is one of participents trying to achive clarity and understanding of the ideas they expound and, in this thread particularly, add detail and nuance to them in the process.

Thank you for taking the high road and encouraging the serious conversation we have enjoyed on this thread.

Now something I do have an area of expertise in is the venues of New York City. Avery Fisher Hall is best for theatre not music. Full of dead spots and more bounce then a pensy pinky. It is a failed experiment that an acoustic remodling failed to cure. The New York State Theatre and The Met are architectural monuments of beauty but not sound. Carnegie Hall has the BMT train running underneath it (besides practice its the best way to get there).

Now Town Hall is nice, it once housed the NY Philharmonic and Metropolitan Opera. It is a preferred medium sized venue. A while back I saw Elvis Costello and the Brodsky Quartet perform the Julliet Letters there. Very intimate yet a very large sound.

It is a shame that the plans to move the Met and Lincoln Center as part of a downtown arts center rebuilding the Ground Zero Battery Park area never bore fruit. IM Pei or Frank Gehry was supposed to design it.

The Opera Houses of Europe like the Concertgebouw in Amsterdam, the Gustav Mailer Hall in Viena, and La Scala Opera, these are made for Music.

In NYC give me Upstairs at Max's, CBGB's, the Fillmore East, the Lennox Lounge, The Blue Note, The Village Vanguard,The Ritz, Bonds, Irving Plaza The Cotton Club, the Bottom Line and the Apollo Theatre.

If I had to think of the place I'd like my system to sound like it would be the Electric Circus back in the sixties on Saint Marks Place. When Hendrix was recording across town at Electric Ladyland Studio he would play with everyone there. He once said if I could just get whats in my head out on stage or on the record it would be far out., but I just can't do it man.

BTW Thom did you know that on the Beatles first American tour ticket sales were poor at Red Rocks so they canceled a show, but they sold out the Hollywood Bowl twice, but I have digressed.

Thanks again

Groovey Records

Listening to
Stravinsky-The Firebird-Anatal Dorati-London Symphony Orchestra
Original Mercury Living Presence SR 90226

Groovey one ...

Yes, it goes without saying that halls, mikes (and miking techniques) can drastically alter the recorded sound for good or for ill.

I was on the phone with a recording engineer today and we were waxing philosophically about how under-emphasized pro-sound (recording) practices are in hi-end audio.

I have greatest respect for the opinions of Ralph Karsten, Jim Hagerman, Victor Khomenko, and Nick Doshi.

Having said that, Jim's and Victors' and Nick's arguments about balanced operation in a home audio context make more sense to me than Ralph's do.

Nick Doshi, for example works in a broadcast environment but in this case advocates a deviation from pro-sound practice by advocating single-ended phono stage operation.

What's to be concluded from this? If it sounds good, it IS good.

Sorry for the relativism here, but at the end of the day the only meter that you should be concerned with is the one that measures the width of the smile on your face.

It's we designers who have to sweat the details and numbers.

Now, as far as balanced is concerned, you can only try it and report back. Theory is just that and nothing more.

I have no doubt Ralph's experiences with balanced hookups are real and I would advise someone with an Atmasphere RIAA stage to experiment with a balanced hook-up.

It's Ralph's explanation that makes no sense to me however, as well how relevant it is to other balanced gear like Ayre, Hagerman, and BAT.

Please bear in mind that a correlation (Balanced hook-up in product X sounds good) does not imply causality (all balanced hookups are good), but at least you'll know what works for you in your situation.

Cheers,
Thom @ Galibier.

Hi Thom, what is it about my explanation that you do not understand?

What impressed me about talking to Nick (among other things) was that we both came from the pro audio background and he seemed very sensible about his design issues with hi fi gear. He doesn't throw around a lot of snake oil.

One thing I can tell you about tape machines and recording studios where all this music we listen to is made, is that power supplies have long been the devil. This may not be news to anyone but if get into the pro audio community you'll find slathering comments about certain ampex atr's and never boards and 3m half inch two tracks. In La, where i lived for a long time, most of the good studios like capitol, had dean jensen go through and rebuild everything. Their atr's machines were not nearly stock. not were the consoles at olympic or the neve's at ocean way or really any piece of gear that was famous. And, almost always, the power supply is the big thing that was skimped on.

nick's heavily damped unit, with separate supplies and keen attention to grounding is probably one of the reasons why his stuff sounds so great even though he didn't re-invent the wheel design-wise. I think he just has the diagnostic issues that create problems in other units licked.

I had an aesthetix rhea and the doshi absolutely embarrassed it for noise floor, detail and neutrality. I have not tried everything out there, but I kind of don't want to anymore.

Hi Ralph,

Likely this best discussed (or put to rest) in the already heavily beaten to death thread on balanced inputs to phono stages - where you and Jim H. had an "enthusiastic" discussion - ultimately agreeing to disagree.

Two of the main points I can't get around are:

(1) Whether or not a cartridge is a balanced or a floating single ended device. Floating single ended makes more sense to me.

and

(2) Since everyone more or less agrees that there isn't a 6dB noise advantage to running a cartridge in "balanced" mode, what might the advantages of running a cartridge in balanced mode be.

Given the fact minds far greater than mine disagree on the subject, I'm content to ultimately let my (and your) ears be the final arbiter of goodness.

Cheers,
Thom @ Galibier

I can see why some people don't like revealing systems for rock, but I always prefer neutrality (while still retaining musicality), it can just take some getting used to.

My dunlavy's are ruthless compared to my B&W's and some people think B&W's are overly neutral! The dunlavy's give you ever speaker cone distortion from an amp amd every rattle from a string while the B&W's are a little more cohesive with more of a mid-bass bump. The things that takes getting used to is how some of the older recordings especially certain rock stuff that was mixed on altec 604's, just really changes a lot on a revealing system. But that resolution is also addictive and I find myself not caring what speakers I hear chuck berry on. He's still great and the players are still great. The recordings reveals strange things but to me nothing gets in the way of music except bloaty or harsh colorations. Many tube products do this and I am with doug on not dialing sweetness into the system.

There is a line where a system becomes unmusical, but i find if it is unmusical it is usually not reproducing music the way it was heard by the microphones.

Some people find certain things musical that are really not very neutral and these things strike first and bug you later. wilson speakers come to mind, but to each their own.

Two of the main points I can't get around are:

(1) Whether or not a cartridge is a balanced or a floating single ended device. Floating single ended makes more sense to me.

and

(2) Since everyone more or less agrees that there isn't a 6dB noise advantage to running a cartridge in "balanced" mode, what might the advantages of running a cartridge in balanced mode be.

Hi Thom, since a cartridge is an inductive device, it has something in common with other inductive devices like audio transformers. Any output of an audio transformer can be used in the balanced mode, even if it is driven single-ended. All that is required is that neither side be at ground. Ground can be the chassis the transformer is on; in the case of a cartridge ground can be the tonearm/turntable.

BTW a 'floating single-ended' device will hum like the dickens. Single ended devices are always grounded.

I have a tube mixer I built that is single-ended for my keyboards and drives a single-ended output transformer that is designed for 600 ohms. The output of the transformer is tied to pin 2 and pin3 of the XLR; pin 1 is ground. It drives 600 ohm balanced lines effortlessly.

Cartridges work the same way. It may not have been the intention of a cartridge manufacturer to make a balanced source out of it, but that is in fact how they behave since neither side of the cartridge is 'grounded', i.e. tied to its metal body. In fact many cartridges don't have a metal body! So really the question is more like: how in the hell can this thing be single-ended?" When looked at that way, you suddenly see why there have to be special grounding considerations (ex.: the third grounding wire) that you would not normally expect to see on your typical single-ended output (like from a tuner).

One obvious advantage of operating a cartridge balanced is the interconnect cable itself, which is less likely to have the usual interconnect cable colorations and less propensity for hum/noise pickup. Although the *output* of the cartridge is going to be the same regardless of balanced or single-ended, there is in fact a noise advantage to the input amplifier, simply because it is differential and makes less noise than a single-ended input amplifier.

IOW there is little advantage insofar as the cartridge itself is concerned, but plenty of advantage from everything that you use with it: the cable and the preamp itself.

>>since a cartridge is an inductive device<<

Being inductive should have nothing to do with balance. The transducer could be capacitive (touch sensor) or resistive (thermistor). It's just a two-terminal device.

>>there is in fact a noise advantage to the input amplifier, simply because it is differential and makes less noise than a single-ended input amplifier<<

I don't believe this is true. You get double the gain, but same SNR.

>>there is little advantage insofar as the cartridge itself is concerned, but plenty of advantage from everything that you use with it: the cable and the preamp itself<<

Yes, this is where we agree!

jh

Cartridges work the same way. It may not have been the intention of a cartridge manufacturer to make a balanced source out of it, but that is in fact how they behave since neither side of the cartridge is 'grounded', i.e. tied to its metal body. In fact many cartridges don't have a metal body! So really the question is more like: how in the hell can this thing be single-ended?" When looked at that way, you suddenly see why there have to be special grounding considerations (ex.: the third grounding wire) that you would not normally expect to see on your typical single-ended output (like from a tuner).

That's correct. I just wish every audio designer got an obligatory course on noise theory and balanced systems. There would be much less misinterpretation and mythology about this important advancement. Balanced is one of the greatest ideas in audio history.

Although the *output* of the cartridge is going to be the same regardless of balanced or single-ended, there is in fact a noise advantage to the input amplifier, simply because it is differential and makes less noise than a single-ended input amplifier.

Correct again, provided you are talking about RFI, hum and other sorts of external EMI entering through the input cable as a common mode signal. In MC cartidges, hum rarely disappears completely, but a balanced system will dramatically reduce it compared to a single-ended system. If you mean thermal noise (hiss), the right answer is: It depends of the design of the preamplifier. You can design a balanced circuitry with much less noise than a single ended one. It just depends on your skill and the technology you are using.

Being inductive should have nothing to do with balance. The transducer could be capacitive (touch sensor) or resistive (thermistor). It's just a two-terminal device.

Correct. Strain gauges, which are resistive elements, also work by using the differential principle.

>>there is in fact a noise advantage to the input amplifier, simply because it is differential and makes less noise than a single-ended input amplifier<<

I don't believe this is true. You get double the gain, but same SNR.

Incorrect. Strictly speaking, a balanced circuit will have two input gain cells operating in differential mode. This circuit would produce 3 dB [20log(sqrt(2))] more noise than a single gain cell operating in identical conditions. With double the gain (6 dB), the result would be a net loss of 3 dB in SNR. However, as said above, it depends on the designer's skill and the technology used. There's no limit on how noiseless a circuit can be (balanced or not), except that imposed by nature.

Regards,

This circuit would produce 3 dB [20log(sqrt(2))] more noise than a single gain cell operating in identical conditions

Except the signal is split between the two inputs! This is not the same as two amplifiers operating in parallel, whereupon you would get the 3dB SNR advantage.

jh

raul,

the vast majority of recording engineers to not like uncolored recording equipment in my experience. The ubiquitous sound of the proximity effect on a neumann u-47 up close that bumps in the high midrange and rolls off steeply in the higher freq.is evidence of this. Especially today, accuracy is not the name of the game. heavy editing and the close mic'ing of soloists in classical music, changing the sound stage has taken some of the natural quality that remained in classical music recording longer than some forms, out of it.

The very idea of multiple microphones really changes what the ear would hear "live". And yes, you can out you ears when the mics are, but how the mics are mixed together and their on and off axis response are just two factors of many that make recording an art as much as a science. Sometimes microphones with poor freq response measurably, have more of a "you are there" feeling than people would imagine. Sort of the like quad 57 speaker can be very musical. The famous rca living stereo string sound is wonderful but far wider than an actual orchestral image.

I tend to be digressive in these discussions and I apologize, but given your post on this, I wanted to point out that these things are carefully designed often for the colorations they bring to music. Reproducing equipment hopefully is trying for less coloration and more neutrality in that it gives the recording its intended sound. But recordings themselves are all over the map. Engineers, sadly are moving even further away from reality as they have been doing since the 70's when heavy isolation came into vogue.

Quite so, Mothra. I've been at recording sessions where I was able to physically put my ears where the microphones were, and then listen to the electrical feed from microphones. In the majority of cases, there is a substantial difference between what the ear and microphone hears, even from the same location. And in most modern recordings, a variety of microphones are used, and each modifies the sound in distinctive ways.

Also, the location and angle of the microphones will in most cases be quite different from what you would hear if you were at a performance of the same event. Most microphones are located far closer to the instrument than any audience seat, and the angles will be quite different, too.

If you have a friend who plays the violin (for example), it is very instructive to listen to it being played at a distance (like you would hear from an audience seat), then listen to it from a distance of under one meter to get the microphones' perspective, and also listen to the instrument from above (again to get the microphones' perspective).

Now put all of the above together and think about the implications for a home audio reproduction system. Since the recording likely does not sound like what you would have heard live from a seat in the audience, if you have set up your audio system to sound like what you'd hear live, it is almost certain that your audio system is modifying what's on the recording, and not in a small way, either!

However, there are recordings that include a list of the equipment used, and also microphone placement drawings. If you know what the recording gear sounds like, and also study the placement drawings, you can form a closer guesstimate of what these recordings should probably sound like, and this can be a somewhat better guide to setting up your system (although you still won't know what the mixing contributed, as Mothra pointed out).

yes, and some of those "you are there" recordings like roy dunan's Rollins "way out west" are all u-47's and c-12's. very colored mics with a lot of proximity effect in cardioid pattern. I'm do this for a living and I can't make a record sound that good (nor have i ever had players of that caliber), so i'm not knocking it. But you're right that the live sound and the mic feed is often very different. Sometimes good, sometimes bad.

I have great respect for your ability to make cartidges that bring out those colored sounds with an even hand though!

And my hat goes off to you, and all recording engineers who capture those great performances which justify our having audio systems!

It is really helpful to work with live recording sessions and master tapes, especially if they find their way to LP!

I've done a lot of recording and after a while you get used to the way your mics 'hear' and place them where they will get the effect you are after- not always where you would put your ears.

Hagtech is quite correct that the device does not have to be inductive; IOW a balanced source is certainly not limited to inductive devices (like microphones, transformers and cartridges). You can even get ceramic devices to be balanced, but who would want to- like frying an egg on the sidewalk :)

IMO, careful design of differential balanced circuits includes the use of properly designed constant current sources and you will want to have a properly regulated power supply too. A proper CCS will dramatically increase the differential amplifier's ability to reject power supply noise, and again if done right seems to reduce the overall noise floor of the differential amplifier as well.

I my case where this took me was less stages of gain overall. Figuring out what the tubes needed to be quiet was the key.

I my case where this took me was less stages of gain overall. Figuring out what the tubes needed to be quiet was the key.

Ralph, It sounds like you and my friend Mr. Doshi are working from similar briefs, and from experience I know that's a good thing. I haven't heard your preamps but I think I'd like them.

Mothra,
It's great to see you posting here. Thom's thread has collected an impressive group of designers, plus some lucky listeners like you and me.

Your comments on cohesiveness vs. transparency nicely outlined the divide between the majority of rock listeners (I think) and a minority - like you and me - who prefer transparency to the source even for that genre. For better or worse, my ears usually won't let me enjoy a less transparent component once I've heard a more transparent one. I can't stop hearing temporal smear or overhang as an artifact, even if it blends some music or recordings into the appearance of a more seamless whole.

I recognize I'm in the minority on this, at least among rock listeners. It has occasionally gotten me into trouble with them, since we hear or respond so differently. Even Paul, whose ears are faster than mine, prefers a more cohesive sound for rock.

BTW, which B&W's do you have? I recently tried something on my N803's that I hadn't bothered trying for a couple of years - removing the tweeter screens. We've always listened without the mid/bass grilles, but when I removed the tweeter screens two years ago the sound did not improve. IIRC it sounded more shrill. That didn't make sense to me at the time, but I now understand I had HF problems (including smearing) with upstream components and wire. Removing the screens was letting me hear those more clearly.

Our upstream components are now *much* better, including Nick's preamp, amp and nearly everything else. Removing the tweeter screens now provides a large improvement. HF's are clearer and less edgy, imaging/soundstaging are much improved, just what you'd expect from removing all that diffracting wire mesh. Bass clarity has improved too, which I find weird. People report that supertweeters give a surprising boost to bass quality and removing our tweeter screens has had a similar effect. Try it if you haven't.

Sorry for the digression folks. Back to the preamp-a-thon.
Doug

Thanks for the kind words, Doug. I tend to be digressive in my posts, so I don't want to mess up the threads when real designers are talking!
I have the B&W 801 matrix series 3 limited edition(which i think is just the finish and the fact they have a plaque that says "abbey road" on them). While the nautilus series smoothed out the tweeter, I find the last matrix series more brutally honest about the treble if not as flattering. I think maybe the matrix series got a little bit of a bad rep because the versions before had protection circuits as they were used in studios and it affected the sound. I thought the last series were great. they have a little lower mid-bass bloom that makes them a little more cohesive for rock and roll than say dulavy's on in the extreme, martin logans (a different animal!).

I just like hearing the details. I think if I didnt have a tube amp maybe the dunlavy's wouldn't see as much action. I also have quad's and spendors which are less precise but very musical. But one gets addicted to the details good or bad, and for me the bad stopped bother me long ago.

My choice of speakers has a lot to do with my work. If I listen to a mix I did and it stands up on the dunlavy's and the 801's then it will generally stand up on "friendier" speakers.

I work cheap and generally in very primitive environments and i often (gasp) use "color" in my recordings. But i want to know what I am doing wrong!

The doshi comments are right on. everything improved in my system and seemed to be more neutral without being to clinical.Or maybe I like clinical I just don't like harsh and thin. Who knows.

"Since the recording likely does not sound like what you would have heard live from a seat in the audience, if you have set up your audio system to sound like what you'd hear live, it is almost certain that your audio system is modifying what's on the recording, and not in a small way, either!"

Thank you, Jcarr, for stating this so succinctly! As a fellow recording engineer, I have often rolled my eyes at the otherwise authoritative expositions touting a components ability to sound like "The Absolute Sound" of live music. Recordings themselves, often sound like anything but. The bottom line is that it is all relative, although there is some utility in aspiring to neutrality and true transparancy at each step. It can be a elusive path to aspire to, though. With that in mind, there are some recordings that do effectively aspire to natural accuracy, and can, as such, be used as a reference for dialing in a system to sound like live music, as well as exhibiting fidelity to the source, whatever it might sound like.

... and thank you all for being so willing to discuss the inner workings of these issues so candidly. I am both encouraged and discouraged to find the mystery persisting despite all our efforts. Viva la difference!

Hi all,

Thanks Jonathan, Mothra, Piedpiper and others for adding some very needed focus and perspective on the variability of the recording process, microphone selection, monitoring, etc. etc.

Ultimately, as Piedpiper stated, we need a small cache of recordings we can rely on to help us map out reality and point the way to the musical truth.

I've found that this doesn't necessarily demand that we depend on unobtanium records from the golden age of recording, although this certainly helps.

You can approximate this by listening to your RECORD COLLECTION ... no small task, I realize.

Now, I don't mean listening to your entire collection, but at the same time, realize that you can effectively interpolate your results by intelligently sampling your collection ... filling in the dots if you will.

You should be reasonably familiar with a collection of recordings - ones which duplicate some well known sounds - be it a tenor, a soprano, a Strad, some drums, a Strat (played through a Tweed Deluxe amp for example).

Now, that SAME tenor will sound completely different depending on the engineer, the mastering, the mike selected, etc., but if you listen to a dozen different records, you'll be able to find a commonality amongst them that will serve you well.

Add to this, your soprano, some strings, horns, drums, and a gee-tahr, and this process of triangulating (a dirty word in politics these days, but helpful in navigating this maze) on reality will take you home.

Is this process perfect? Nope. Will it help you get the most out of your record collection? Yup.

One problem with this method is that any single change in your system will upset the apple cart. Still, if you follow this process with an open and attentive mind, it can serve you well.

Broadening your "reference" record collection will serve another purpose.

It will shift your focus away from having a "shootout mentality" and will force you to be patient. Likely (hopefully) your attention will return to what's really important - MUSICAL COMMUNICATION.

All too frequently, when we use a few "pet sounds" (sorry ... couldn't resist), we focus on those sounds instead of on music.

The digital crowd stops at this point (sound) wondering what all of this analog mania is all about. How sad for them, and indirectly for us, because being the majority, they dictate their sensibilities to the recording process - but that's a whole 'nuther topic.

Cheers,
Thom @ Galibier

Digital RIAA implementation via DSP/VST:

@ 96 kHz:

two pole IIR filter:
MAXIMUM ERROR FROM 0.00 Hz TO 20000.00 Hz IS 0.0057028dB
MAXIMUM PHASE ERROR FROM 0.00 Hz TO 20000.00 Hz IS ~+/- 2.4 degrees

three pole IIR filter:
MAXIMUM ERROR FROM 0.00 Hz TO 20000.00 Hz IS 0.0000046dB
MAXIMUM PHASE ERROR FROM 0.00 Hz TO 20000.00 Hz IS ~+/- 1.6 degrees

http://www.head-fi.org/forums/f46/software-based-riaa-eq-part-2-a-311909/index2.html

Doctor C ...

Give us some system context on this "perfect RIAA forever".

Cheers,
Thom @ Galibier

Thom

Give us some system context on this "perfect RIAA forever"

I'm not the Dr but, were you to try digital (for archiving purposes I assume), I would recommend you use the 192kHz version. The system is convoluted &, of course, makes no sense for casual listening: feed TT analogue signal to puter, convert to digital, digital deemphasis, reconvert filtered digital result to analogue... whew!

The actual equalizer works fine as mentioned above; it's the sound that's horrible!

Years ago the Library of Congress did a study; they concluded that any laminated media was not/is not servicable as an archival media. IOW any laminated media would fail within years or 2 decades at the most.

Non-laminated (amorphous) media is preferred as it cannot delaminate over time. So far the only examples of this sort of media are LPs and 78s, and the stampers that made them. LPs have a projected storage life that goes into centuries if properly stored!

I can simultaneoulsy monitor AND record the digital RIAA setup with 11% cpu usage. It is not convoluted.

I don't believe he was talking about CPU usage.

I don't have the patience to read this whole thread, so this may already have been addressed: high capacitance PSs charge "faster", not slower than low capacitance PSs. That is, unless there is some series resistance between them and the rectifiers. High capacitance equates to low impedance. This results in very small conduction angles. The current pulses through the PS transformer in short, high current pulses that cause ringing and generate lots of high freq. harmonics. This is a plus for using minimal capacitance in these PSs. They take longer to charge and draw current though the transformer more smoothly. Both my line stage and phono pre have choke input PSs with minimal capacitance (30uF or less) and rely on cascoded MOSFET CCSs to block the ripple (some of the current supplied to the gain stage is shunted across a resistor to develop the B+ voltage). With the gain stages being differential triode pairs with "stiff" CCS "tails", each gain stage is very isolated from the PS. I mention all this because when Kevin Carter had this phono pre in development he had to dispense with the choke input to have enough B+ to work with. Even with CCSs that have an AC impedance >80M ohms the sound was noticably degraded by the capacitive input PS. Lesson here: PS hash is real and can be an issue with low level signals.

That is why stiff regulation and effective Constant Current Sources are a must. IMO, an effective CCS is one that will create immunity to the supply- independent of the fact that the amplification is fully differential.

I just wanted to pass on a PS update. I recently rewired the PS chokes in my line stage and phono pre. Lundahl PS chokes, like all Lundahl chokes, have 2 coils wound on a C core. Previously both coils were connected in series with the B+ side on the rect. bridge. Lundahl shows an alternative connection in their data sheets where one coil is in series with the B+ side and the other in series with the neg. side. It is their claim that this provides better common mode rejection. It definately made for a quieter and more ingaging listening experience. My guess is that it is preventing PS hash from contaminating the circuit ground. This in a line stage and phono pre with cascoded MOSFET CCSs isolating each differential pair on both the ground and B+ side. Truly quiet PSs and grounds seem to be a requirement for good sound.