Looks like you're having fun with the basshorn! My projects are moving in a commercial direction, with a possible introduction this year. Or not, depending on how much progress I make. The measurement system update, with a MacBook Pro laptop dual-booting into OS X and Win7, running ARTA and AudioTester, and using the just-arrived M-Audio ProFire 610 interface, is an important part of this.
I was really out of the loop for several years after the accident, with the main system not even running. About a month before the Rocky Mountain Audio Festival (which is only about 25 miles from where I live), I decided that it was time to get things running again, and this time use the really good modified Denon DV2900 I picked up at one of the Colorado Audio Club meetings as a CD/DVD-A transport.
So I finally got everything going, and to my surprise, it sounded much better than I remembered, with the annoying sonic variability of the Karna amplifiers all gone. A big part of this was the much richer and deeper sound of the new transport - the previous Pioneer cheapie was really quite thin-sounding, with marginal dynamics. A basic quality of the Karna amplifiers is they will not cover up faults in the source; if the source is thin-sounding, the whole system will sound that way. The quad set of TJ meshplates sounded pretty good too; no need to chase after exotic 300B types.
The most gratifying thing of all was the sound quality compared to what I heard at the RMAF; I'm probably prejudiced, but I preferred my own system to anything at the show. That's not surprising; I voice my system for my own tastes, and they're fairly different than the high-end audio mainstream. What I didn't expect were my guests saying the same thing, playing their own selections of music. I felt a lot better about the Ariels and Karna amplifiers after that.
Two of my visitors brought their own state-of-the-art DAC along, and really wanted to compare to the Monarchy, as well as hear it on the Karna/Ariel system. I thought the expensive DAC - at five times the price - would just clobber mine, but to my surprise, I preferred the Monarchy. On every music selection. I just prefer the sound of a Burr-Brown multibit converter with passive I/V conversion and vacuum-tube amplification. More vivid, more colorful, more involving.
The Karna/Ariel combination has such high resolution that chasing detail isn't important; what matters much more is the quality of the source - how much music it lets through. Some of the more analytical and supposedly "neutral" sources really squash the dynamics and tone colors of the music, which flattens out the emotional qualities of the performance.
Which gets back to the phonograph. My tastes really aren't in the mainstream of high-end audio; I frequently - but not always - like the sound of a more vintage approach. The Philips 16-bit DACs are not my cup of tea, even in NOS form with vacuum-tube amplification. I like the Burr-Brown 20-bit converters better, and prefer well-implemented upsampling to non-oversampled Red Book digital. But the delta-sigma converters with solid-state buffer/amplifiers really don't do it for me; maybe they're aimed at the transistor-amp crowd, since they have such a different tonal character. I like vivid, intense, and immediate sound, and the Karna amplifiers certainly deliver that, with sonics in the WE 212E (not 211) class.
I've hear all of the exotic cartridges - the $5000 and up Dynavectors and Zyx models at Thom Mackris' place, and I'm not too sure that's the sound I want. Most of my LP's are vintage, and since they were always played on my system (instead of the really gruesome low-fi systems of the day), they're in pretty good shape. Yes, I have a few audiophile demo discs, but very few of the reissues which are probably remastered with digital preview systems and modern Class AB transistor amps driving the cutterhead. Most of these pricey reissues do not interest me, unless they sound a lot better than the originals.
It was hearing Christian Rintelen's system in Zurich that was the transformative experience. Now that was what LP's were supposed to sound like. And he was using the SPU cartridge, not one of the modern audiophile wonders. What surprised me was the SPU-to-Denon 103 comparison; I didn't like the Denon at all; it sounded flat and gray in comparison to the super-vivid tone colors of the SPU.
I am curious about the Audio-Technica OC9/MLIII, but I suspect the SPU is the real treasure. I'm also wondering how the SPU (in the non-headshell version) would sound on the Jelco 750D arm.
I was really out of the loop for several years after the accident, with the main system not even running. About a month before the Rocky Mountain Audio Festival (which is only about 25 miles from where I live), I decided that it was time to get things running again, and this time use the really good modified Denon DV2900 I picked up at one of the Colorado Audio Club meetings as a CD/DVD-A transport.
So I finally got everything going, and to my surprise, it sounded much better than I remembered, with the annoying sonic variability of the Karna amplifiers all gone. A big part of this was the much richer and deeper sound of the new transport - the previous Pioneer cheapie was really quite thin-sounding, with marginal dynamics. A basic quality of the Karna amplifiers is they will not cover up faults in the source; if the source is thin-sounding, the whole system will sound that way. The quad set of TJ meshplates sounded pretty good too; no need to chase after exotic 300B types.
The most gratifying thing of all was the sound quality compared to what I heard at the RMAF; I'm probably prejudiced, but I preferred my own system to anything at the show. That's not surprising; I voice my system for my own tastes, and they're fairly different than the high-end audio mainstream. What I didn't expect were my guests saying the same thing, playing their own selections of music. I felt a lot better about the Ariels and Karna amplifiers after that.
Two of my visitors brought their own state-of-the-art DAC along, and really wanted to compare to the Monarchy, as well as hear it on the Karna/Ariel system. I thought the expensive DAC - at five times the price - would just clobber mine, but to my surprise, I preferred the Monarchy. On every music selection. I just prefer the sound of a Burr-Brown multibit converter with passive I/V conversion and vacuum-tube amplification. More vivid, more colorful, more involving.
The Karna/Ariel combination has such high resolution that chasing detail isn't important; what matters much more is the quality of the source - how much music it lets through. Some of the more analytical and supposedly "neutral" sources really squash the dynamics and tone colors of the music, which flattens out the emotional qualities of the performance.
Which gets back to the phonograph. My tastes really aren't in the mainstream of high-end audio; I frequently - but not always - like the sound of a more vintage approach. The Philips 16-bit DACs are not my cup of tea, even in NOS form with vacuum-tube amplification. I like the Burr-Brown 20-bit converters better, and prefer well-implemented upsampling to non-oversampled Red Book digital. But the delta-sigma converters with solid-state buffer/amplifiers really don't do it for me; maybe they're aimed at the transistor-amp crowd, since they have such a different tonal character. I like vivid, intense, and immediate sound, and the Karna amplifiers certainly deliver that, with sonics in the WE 212E (not 211) class.
I've hear all of the exotic cartridges - the $5000 and up Dynavectors and Zyx models at Thom Mackris' place, and I'm not too sure that's the sound I want. Most of my LP's are vintage, and since they were always played on my system (instead of the really gruesome low-fi systems of the day), they're in pretty good shape. Yes, I have a few audiophile demo discs, but very few of the reissues which are probably remastered with digital preview systems and modern Class AB transistor amps driving the cutterhead. Most of these pricey reissues do not interest me, unless they sound a lot better than the originals.
It was hearing Christian Rintelen's system in Zurich that was the transformative experience. Now that was what LP's were supposed to sound like. And he was using the SPU cartridge, not one of the modern audiophile wonders. What surprised me was the SPU-to-Denon 103 comparison; I didn't like the Denon at all; it sounded flat and gray in comparison to the super-vivid tone colors of the SPU.
I am curious about the Audio-Technica OC9/MLIII, but I suspect the SPU is the real treasure. I'm also wondering how the SPU (in the non-headshell version) would sound on the Jelco 750D arm.
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As a followup, the eventual preamp may echo some of the features of the Amity and Karna amplifiers, with fully-balanced and transformer-coupled circuits, and possibly LCR equalization. Like the amplifiers, I am more concerned about slew rate, overload, instantaneous recovery, and symmetry than the lowest-possible static noise figure.
Moving-coil cartridges, with their very low inductance and low tip mass, create very fast rise-times whenever they mistrack, which is a fair percentage of the time. Looking at the output of a MC cartridge on a fast analog scope is a sobering experience; those little blips are moving fast, and many, if not most, RIAA phono stages run out of current when these fast-moving glitches come along. The problem area is the first amplifying stage looking into the reactive load of the RIAA network, which can present a very low impedance in the 5 kHz and above region.
A high slew rate into a reactive load will overwhelm many preamps, and the recovery time will be much longer than the original sub-millisecond transient. If the power supply uses active regulation, it might be always bouncing up and down as the preamp stages move in and out of current saturation.
I should add that mistracking in the audiophile sense - of hearing distortion and crud - is very different than looking at a scope. On the scope, you'll see that little glitches are happening all the time, even on a record that sounds very clean. When mistracking gets to the point of being audible, the scope trace is a scrambled mess, and it is very obvious that the stylus is spending a lot of time slamming back from one groove wall to another.
The grossness of mistracking is the strongest argument possible for minimizing tip mass (which is completely unrelated to stylus compliance). The lower the tip mass (the effective dynamic tip mass, not the weight of the diamond), the better the records will survive being played, and the better the cartridge will sound.
Cantilever flex doesn't help matters, since cantilever resonances can easily be engraved directly into the record grooves - permanently. This is a function of peak pressures and contact area of the stylus, with inaccurately manufactured (and mounted) ellipticals being the worst record-eaters. This isn't a theoretical problem: I've seen for myself a cantilever that was misaligned by 10 degrees on microscopic examination - on a $500 audiophile cartridge.
To accurately align that particular cartridge so the stylus fits the groove, it would have to be canted 10 degrees in the headshell. If the stylus - particularly a fine-line type - doesn't fit the groove, there will be very high distortion as the record is destroyed.
I was dismayed to find that 1~3 degree misalignment of the cantilever is actually the industry standard, which means that all cartridges must be azimuth aligned by ear (or measurement) on a mono record - and that a correctly azimuth-aligned cartridge may look noticeably off-tilt as seen from the front. If the cartridge (as a whole) is visibly level with the record, odds are good that the stylus is not fitting the groove, and record wear and mistracking are accelerated.
With a conical stylus, of course, this is not a problem, but it is a problem with an elliptical, and far worse with a fine-line or Shibata profile, which require extreme accuracy in setting azimuth. Fortunately, once set, you don't need to mess with it until you replace the cartridge.
VTA and tracking force affect the sound, of course, but the risk of record destruction is not as severe, unless you lighten tracking pressure to the point where mistracking begins. It goes without saying that mistracking is far more destructive to the grooves than a high (static) tracking force, since mistracking is actually the stylus bouncing from groove to groove at extremely high accelerations and pressures.
So a decent-quality 3D microscope, along with a digital stylus-pressure gauge, is yet another must-have gizmo, along with the inevitable record-cleaning machine. But that's analog for you; tape decks need the heads degaussed and the tape-guides cleaned, as well as checking belt and brake tensions, replacing old electrolytic caps, etc.
Moving-coil cartridges, with their very low inductance and low tip mass, create very fast rise-times whenever they mistrack, which is a fair percentage of the time. Looking at the output of a MC cartridge on a fast analog scope is a sobering experience; those little blips are moving fast, and many, if not most, RIAA phono stages run out of current when these fast-moving glitches come along. The problem area is the first amplifying stage looking into the reactive load of the RIAA network, which can present a very low impedance in the 5 kHz and above region.
A high slew rate into a reactive load will overwhelm many preamps, and the recovery time will be much longer than the original sub-millisecond transient. If the power supply uses active regulation, it might be always bouncing up and down as the preamp stages move in and out of current saturation.
I should add that mistracking in the audiophile sense - of hearing distortion and crud - is very different than looking at a scope. On the scope, you'll see that little glitches are happening all the time, even on a record that sounds very clean. When mistracking gets to the point of being audible, the scope trace is a scrambled mess, and it is very obvious that the stylus is spending a lot of time slamming back from one groove wall to another.
The grossness of mistracking is the strongest argument possible for minimizing tip mass (which is completely unrelated to stylus compliance). The lower the tip mass (the effective dynamic tip mass, not the weight of the diamond), the better the records will survive being played, and the better the cartridge will sound.
Cantilever flex doesn't help matters, since cantilever resonances can easily be engraved directly into the record grooves - permanently. This is a function of peak pressures and contact area of the stylus, with inaccurately manufactured (and mounted) ellipticals being the worst record-eaters. This isn't a theoretical problem: I've seen for myself a cantilever that was misaligned by 10 degrees on microscopic examination - on a $500 audiophile cartridge.
To accurately align that particular cartridge so the stylus fits the groove, it would have to be canted 10 degrees in the headshell. If the stylus - particularly a fine-line type - doesn't fit the groove, there will be very high distortion as the record is destroyed.
I was dismayed to find that 1~3 degree misalignment of the cantilever is actually the industry standard, which means that all cartridges must be azimuth aligned by ear (or measurement) on a mono record - and that a correctly azimuth-aligned cartridge may look noticeably off-tilt as seen from the front. If the cartridge (as a whole) is visibly level with the record, odds are good that the stylus is not fitting the groove, and record wear and mistracking are accelerated.
With a conical stylus, of course, this is not a problem, but it is a problem with an elliptical, and far worse with a fine-line or Shibata profile, which require extreme accuracy in setting azimuth. Fortunately, once set, you don't need to mess with it until you replace the cartridge.
VTA and tracking force affect the sound, of course, but the risk of record destruction is not as severe, unless you lighten tracking pressure to the point where mistracking begins. It goes without saying that mistracking is far more destructive to the grooves than a high (static) tracking force, since mistracking is actually the stylus bouncing from groove to groove at extremely high accelerations and pressures.
So a decent-quality 3D microscope, along with a digital stylus-pressure gauge, is yet another must-have gizmo, along with the inevitable record-cleaning machine. But that's analog for you; tape decks need the heads degaussed and the tape-guides cleaned, as well as checking belt and brake tensions, replacing old electrolytic caps, etc.
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To return to the music server discussion, yes, I saw the review of the Berkeley Audio Design USB-to-S/PDIF converter over at Computer Audiophile e-magazine. Looks fantastic, right? And despite the price, yes, I'm very tempted to just go ahead and buy it. It's always a good sign when the reviewer plunks down the money and keeps the review sample. True, reviewers get a price break, but it isn't as much as you might think. I'd guess he still wrote a check for well over a thousand dollars so he could keep the review sample.
But ... the reviewer is using an all-transistor system, with speakers that I've listened to at length and did not like at all. I like Spectral electronics, and have great respect for the team that designed them. But they're not for me: I'm firmly in the zero-feedback vacuum-tube camp, and that's where I'm staying.
So, reading the review, the system is quite different than my own tastes. I don't care for the speakers, and am indifferent about the electronics. If I were sitting in the reviewer's living room right now, I probably wouldn't be able to make any kind of assessment of the system as a whole, since it would have such a different presentation compared to what I'm used to. Not good, not bad, just not what I listen to.
What that really tells me, aside from my sonic prejudices - and they're just that, prejudices - is the reviewer is listening for different things than I do. This is an important data point - inferred, true - but I think still valid. If the reviewer had the perceptions, tastes, preferences, etc. similar to mine, the speakers and electronics would be different. Not all-transistor. Different kinds of speakers with different voicing. And so on.
This is what hifi is all about. Different people really hear different things. The project manager at ESS Technology really can't hear the difference between delta-sigma converters and ladder converters, but was willing to spend several years and a substantial R&D budget to validate the perceptions of the few staffers and outsiders who could reliably distinguish between the two types of converters - and to improve the performance of the Sabre 9018 so it could no longer be distinguished from a ladder converter.
That's putting your money where your mouth is. He couldn't hear it for himself - he said so right at the beginning of the RMAF presentation - but dug deeply into the properties of delta-sigma converters to find the sonic differences. And they found them - very subtle, but repeatably measurable - and addressed them, one by one. When the fifth parameter was corrected, then, and only then, did the new ESS Sabre converter finally match, and potentially exceed, the ladder converters. I was very impressed that ESS, the inventor of delta-sigma converters, rather than push a propaganda campaign against the rival (and rapidly obsoleting) technology, carried out a substantial R&D program to investigate just what the "Golden Ears" (his words, not mine) were hearing - and to address the deficiencies of their own conversion technology. Can you imagine Sony or Philips doing that?
This is a very different approach than simply dismissing the perceptions of other listeners. There are people out there on the Net - some of them distinguished engineers - who are firmly convinced that lossy compression (at 320 kbits/s rates) is indistinguishable not only from uncompressed 44.1/16, but a live mike-feed. There are others that maintain that 44.1/16 PCM is indistinguishable from a mike-feed, and the Red Book standard is good enough for all time. (I should add for younger readers that the Sony/Philips 44.1/16 system was bitterly criticized at its introduction as being inadequate for consumer and professional use, which is partially why the movie industry settled on the 48/16 standard, which is hard-coded into Dolby Digital/AC3. The pre-existing 50/16 standard, in use by SoundStream and Denon, was set aside.)
I don't agree with the low-bitrate crowd. I hear things other people don't. But other people hear things I don't. I've met people who are extremely sensitive to the absolute phase of a given recording, and flip the phase (of both channels) for every recording they listen to. On some recordings, yes, I can hear a small timbral change. But it isn't very large; I could walk out of the room and back again and forget the difference, or just sit in a different location, and that would be a larger change. So it's something I'm not very sensitive to. I'm not that into cables, either; give me industrial Litz-wire in a cotton jacket and I'm happy. (Thanks, Gary Pimm and Bud Purvine!)
Different people hear different things; that's just how it is. Going further, these different perceptions affect the system they end up listening to; and the choice of a reviewer's system is an indirect indication of their musical tastes and perceptions. There's no good or bad here: it might sound bad to me, but that doesn't mean the reviewer is wrong. They just hear different things than I do.
But ... the reviewer is using an all-transistor system, with speakers that I've listened to at length and did not like at all. I like Spectral electronics, and have great respect for the team that designed them. But they're not for me: I'm firmly in the zero-feedback vacuum-tube camp, and that's where I'm staying.
So, reading the review, the system is quite different than my own tastes. I don't care for the speakers, and am indifferent about the electronics. If I were sitting in the reviewer's living room right now, I probably wouldn't be able to make any kind of assessment of the system as a whole, since it would have such a different presentation compared to what I'm used to. Not good, not bad, just not what I listen to.
What that really tells me, aside from my sonic prejudices - and they're just that, prejudices - is the reviewer is listening for different things than I do. This is an important data point - inferred, true - but I think still valid. If the reviewer had the perceptions, tastes, preferences, etc. similar to mine, the speakers and electronics would be different. Not all-transistor. Different kinds of speakers with different voicing. And so on.
This is what hifi is all about. Different people really hear different things. The project manager at ESS Technology really can't hear the difference between delta-sigma converters and ladder converters, but was willing to spend several years and a substantial R&D budget to validate the perceptions of the few staffers and outsiders who could reliably distinguish between the two types of converters - and to improve the performance of the Sabre 9018 so it could no longer be distinguished from a ladder converter.
That's putting your money where your mouth is. He couldn't hear it for himself - he said so right at the beginning of the RMAF presentation - but dug deeply into the properties of delta-sigma converters to find the sonic differences. And they found them - very subtle, but repeatably measurable - and addressed them, one by one. When the fifth parameter was corrected, then, and only then, did the new ESS Sabre converter finally match, and potentially exceed, the ladder converters. I was very impressed that ESS, the inventor of delta-sigma converters, rather than push a propaganda campaign against the rival (and rapidly obsoleting) technology, carried out a substantial R&D program to investigate just what the "Golden Ears" (his words, not mine) were hearing - and to address the deficiencies of their own conversion technology. Can you imagine Sony or Philips doing that?
This is a very different approach than simply dismissing the perceptions of other listeners. There are people out there on the Net - some of them distinguished engineers - who are firmly convinced that lossy compression (at 320 kbits/s rates) is indistinguishable not only from uncompressed 44.1/16, but a live mike-feed. There are others that maintain that 44.1/16 PCM is indistinguishable from a mike-feed, and the Red Book standard is good enough for all time. (I should add for younger readers that the Sony/Philips 44.1/16 system was bitterly criticized at its introduction as being inadequate for consumer and professional use, which is partially why the movie industry settled on the 48/16 standard, which is hard-coded into Dolby Digital/AC3. The pre-existing 50/16 standard, in use by SoundStream and Denon, was set aside.)
I don't agree with the low-bitrate crowd. I hear things other people don't. But other people hear things I don't. I've met people who are extremely sensitive to the absolute phase of a given recording, and flip the phase (of both channels) for every recording they listen to. On some recordings, yes, I can hear a small timbral change. But it isn't very large; I could walk out of the room and back again and forget the difference, or just sit in a different location, and that would be a larger change. So it's something I'm not very sensitive to. I'm not that into cables, either; give me industrial Litz-wire in a cotton jacket and I'm happy. (Thanks, Gary Pimm and Bud Purvine!)
Different people hear different things; that's just how it is. Going further, these different perceptions affect the system they end up listening to; and the choice of a reviewer's system is an indirect indication of their musical tastes and perceptions. There's no good or bad here: it might sound bad to me, but that doesn't mean the reviewer is wrong. They just hear different things than I do.
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Your remarks about the Ortofon SPU LP playback cartridge reminded me of my own battles with high inductance moving magnet designs, which I abandoned in disgust apart from concluding that some Grado designs into 15k were better than most. 
I suppose that the main design goal of an amplifier for a pickup cartridge is to get a well-behaved purely resistive input impedance which avoids calamities when feedback falls flat on its face. Balanced is nice, of course, but maybe not essential. Transformers seem to be a nuisance, but I could live with one if its main function is to match the load to the cabling and termination as well as possible. It is very hard to get away from a cable impedance of 50-600 ohms, and you really don't want to be terminating that in anything like 47k because that creates a capacitive component with the mismatch.
Because Morgan Jones is an engineer who can explain things simply, I tend to think he is someone who really understands stuff too, so I was interested in his idea of using a matching Zobel on a transformer for Ortofon cartridges to avoid ringing:
Sowter Type 8055
For sure, you should mount the preamp in the turntable ideally, but for all that, a 75 ohm cable terminated in 75 ohm resistance ought to be well-behaved. Now you just need a well-behaved buffer amp at the amplifier input. I can't quite remember the details of my moving-magnet efforts, but I think I sacrificed a bit of noise and used fast simple single transistor emitter followers in front of this sort of classic differential amplifier topology which has ideal common-mode rejection:
None of this is to say that any of my particular notions will apply to your design, especially a class A/B op-amp which is a horror really, but the answer always lies in asking the question clearly, n'est pas?
I suppose that the main design goal of an amplifier for a pickup cartridge is to get a well-behaved purely resistive input impedance which avoids calamities when feedback falls flat on its face. Balanced is nice, of course, but maybe not essential. Transformers seem to be a nuisance, but I could live with one if its main function is to match the load to the cabling and termination as well as possible. It is very hard to get away from a cable impedance of 50-600 ohms, and you really don't want to be terminating that in anything like 47k because that creates a capacitive component with the mismatch.
Because Morgan Jones is an engineer who can explain things simply, I tend to think he is someone who really understands stuff too, so I was interested in his idea of using a matching Zobel on a transformer for Ortofon cartridges to avoid ringing:
Sowter Type 8055
For sure, you should mount the preamp in the turntable ideally, but for all that, a 75 ohm cable terminated in 75 ohm resistance ought to be well-behaved. Now you just need a well-behaved buffer amp at the amplifier input. I can't quite remember the details of my moving-magnet efforts, but I think I sacrificed a bit of noise and used fast simple single transistor emitter followers in front of this sort of classic differential amplifier topology which has ideal common-mode rejection:
An externally hosted image should be here but it was not working when we last tested it.
None of this is to say that any of my particular notions will apply to your design, especially a class A/B op-amp which is a horror really, but the answer always lies in asking the question clearly, n'est pas?
A balanced floating circuit is certainly the way to go for a low-level phono cartridge. Once RFI, hum, TV sync buzz, and other forms of noise get into the preamp, there's no way to get it out.
I like transformers because they both filter off RFI and break the ground connection, solving a lot of problems with ground-loops between components. And playing with impedance matching with the cartridge, including pole-zero cancellation at the top of the band, is always a good idea. Test records come in handy for this kind of experimentation.
I even think a 1:1 transformer is a good idea for MM or MI cartridges; as before, it filters off RFI, breaks the ground connection between the TT and the preamp, and offers a balanced connection between cartridge pins and the input stage of the preamp. Whether the internals of the preamp are balanced, or not, is a separate question from the optimum connection between the cartridge pins and the input grid/base/gate of the preamplifier.
An unbalanced RCA connection is just about the worst connection possible, with many opportunities for ground loops, RFI incursion (almost a certainty in many locations), and many annoying choices for "grounding" the turntable, arm, and other metallic bits and pieces of the record player.
How much better to simply float and balance the circuit from cart pins to preamp; the other "grounds" don't matter at all at that point, since they fully isolated from the input circuit.
I've watched these poor audiophiles spend hours fiddling around with various ways to "ground" their tables while knowing that any good-quality isolation transformer offers 60 to 80 dB of noise isolation. The RCA connection is just stupid; high-quality tonearms and preamps should have offered studio-grade XLR balanced connections for the last several decades. An RCA connection (for a high-output ceramic cartridge) made sense at the back of a Zenith table radio in 1956; it made no sense at all for a moving-coil cartridge in 1976, and it hasn't improved in the last three decades, despite cable costs rising to insane levels.
When I see a $3000 cable terminated with RCA connections it reminds me of a turbo V8, overhead cams, and variable valve timing powering a Schwinn bicycle. Something is wrong with the concept.
I like transformers because they both filter off RFI and break the ground connection, solving a lot of problems with ground-loops between components. And playing with impedance matching with the cartridge, including pole-zero cancellation at the top of the band, is always a good idea. Test records come in handy for this kind of experimentation.
I even think a 1:1 transformer is a good idea for MM or MI cartridges; as before, it filters off RFI, breaks the ground connection between the TT and the preamp, and offers a balanced connection between cartridge pins and the input stage of the preamp. Whether the internals of the preamp are balanced, or not, is a separate question from the optimum connection between the cartridge pins and the input grid/base/gate of the preamplifier.
An unbalanced RCA connection is just about the worst connection possible, with many opportunities for ground loops, RFI incursion (almost a certainty in many locations), and many annoying choices for "grounding" the turntable, arm, and other metallic bits and pieces of the record player.
How much better to simply float and balance the circuit from cart pins to preamp; the other "grounds" don't matter at all at that point, since they fully isolated from the input circuit.
I've watched these poor audiophiles spend hours fiddling around with various ways to "ground" their tables while knowing that any good-quality isolation transformer offers 60 to 80 dB of noise isolation. The RCA connection is just stupid; high-quality tonearms and preamps should have offered studio-grade XLR balanced connections for the last several decades. An RCA connection (for a high-output ceramic cartridge) made sense at the back of a Zenith table radio in 1956; it made no sense at all for a moving-coil cartridge in 1976, and it hasn't improved in the last three decades, despite cable costs rising to insane levels.
When I see a $3000 cable terminated with RCA connections it reminds me of a turbo V8, overhead cams, and variable valve timing powering a Schwinn bicycle. Something is wrong with the concept.
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I don't mind RCA phono, as long as it's terminated in (what?) 75 ohms.
But it never is...
Life would be easier if cartridges had a centre-tap too. But they don't.
I seem to recall some very satisfactory sounds from moving coils driving a 100 ohms or so into a common-base pre-preamplifier through a phono lead.
My own experiments suggested the behaviour of the cartridge in the groove was very strongly affected by the load. A low resistance load seemed to increase vinyl-wear too, because the stylus effectively has to be pushed harder to move. That's kinda obvious.
Anyway, Lynn, sounds like your on it. Always enjoy reading your stuff.
But it never is...
Life would be easier if cartridges had a centre-tap too. But they don't.
I seem to recall some very satisfactory sounds from moving coils driving a 100 ohms or so into a common-base pre-preamplifier through a phono lead.
My own experiments suggested the behaviour of the cartridge in the groove was very strongly affected by the load. A low resistance load seemed to increase vinyl-wear too, because the stylus effectively has to be pushed harder to move. That's kinda obvious.
Anyway, Lynn, sounds like your on it. Always enjoy reading your stuff.
Various
Interesting stuff, plenty to think about there - some of which I knew !
At the end of the day, it's worth saying that despite all the potential problems, it's not that hard to get vinyl to work and to sound better than CD ! If you described the process to someone who had never heard of analogue reproduction, you would be laughed out of the room, it sounds absurd ... but it can sound glorious . In my experience, the problems with noise (/hum ) and ringing from step-up inductances can be overstated . I find there are bigger problems is choice of cartridge ( 'HiFi' vs 'music' cartridges ) than some of the subtleties of how you get the music into the phono .
A few personal observations :
103R sounds significantly better than 103 , I have tried 103 again recently and can't get enough tonal colour out of it, with any loading or arm mass, step-ups or jFET phono front end .
The SPU is another step up and very tough to beat with anything .
The 103R can sound excellent with a non-resonant arm and extra headshell mass. It works surprisingly well on a Mayware formula IV unipivot , with extra headshell mass. A good 12" arm is really the way to go though - maybe Thomas Schick's arm, which I'm told by a good friend who owns one, is superb with the SPU .
I've used ZYX R-1000 , Koetsu Red, Dynavector XX-1L and Dynavector XX-2 II in my system, and prefer the 103R to all of those. There are very few really 'Musical' cartridges out there. I'm told the EMT carts are superb, and maybe the Allaerts would be too , both rather expensive. There are very few carts that can represent the tone and impact of a piano properly, the 103R is one of the few.
The only CD player I've ever heard that had some of the essential properties of vinyl was an Audio Note combo : transport '1' and DAC 2 , heard at a HiFi show in Manchester. This is the only time I've heard natural treble with tone - normally CD has a sort of 'crunchy' or 'pixellated' treble , to my ears, that sounds very unnatural , especially on guitars .
I think the AN stuff tends to use the AD1865 , is that Burr-Brown ?
Although step-ups are very good in most respects, and help on the noise front, it has recently been shown to me how much they lose in the low bass and timing , presumably due to primary inductance issues . This thread :
audio-talk :: View topic - j-FET / Triode Phono front-end
Documents my experiments in building a jFET/triode cascode front-end for the phono, and the trials and tribulations of getting it to work . You will notice Joe Roberts and JC Morrison making an appearance in there, as some of JC's ideas ( see 'Lab JC' ) were being discussed.
At the end of the day, I haven't been able to get the jFET to work to my satisfaction yet, but it's shown me that there is a lot more to the low bass rendition than people using step-ups might realise. The change was NOT subtle !
However, the use of a 103 or 103R into step-ups is a sort of worst case really - the 103 being 40R and the 103R being 12R internal impedance. That tends to suggest the SPU should be the choice here , as it will have somewhere from 6R to 2R Zout , which should sort-out the bass issues to a large extent .
Anyway, that's enough of my ramblings ....
Interesting stuff, plenty to think about there - some of which I knew !
At the end of the day, it's worth saying that despite all the potential problems, it's not that hard to get vinyl to work and to sound better than CD ! If you described the process to someone who had never heard of analogue reproduction, you would be laughed out of the room, it sounds absurd ... but it can sound glorious . In my experience, the problems with noise (/hum ) and ringing from step-up inductances can be overstated . I find there are bigger problems is choice of cartridge ( 'HiFi' vs 'music' cartridges ) than some of the subtleties of how you get the music into the phono .
A few personal observations :
103R sounds significantly better than 103 , I have tried 103 again recently and can't get enough tonal colour out of it, with any loading or arm mass, step-ups or jFET phono front end .
The SPU is another step up and very tough to beat with anything .
The 103R can sound excellent with a non-resonant arm and extra headshell mass. It works surprisingly well on a Mayware formula IV unipivot , with extra headshell mass. A good 12" arm is really the way to go though - maybe Thomas Schick's arm, which I'm told by a good friend who owns one, is superb with the SPU .
I've used ZYX R-1000 , Koetsu Red, Dynavector XX-1L and Dynavector XX-2 II in my system, and prefer the 103R to all of those. There are very few really 'Musical' cartridges out there. I'm told the EMT carts are superb, and maybe the Allaerts would be too , both rather expensive. There are very few carts that can represent the tone and impact of a piano properly, the 103R is one of the few.
The only CD player I've ever heard that had some of the essential properties of vinyl was an Audio Note combo : transport '1' and DAC 2 , heard at a HiFi show in Manchester. This is the only time I've heard natural treble with tone - normally CD has a sort of 'crunchy' or 'pixellated' treble , to my ears, that sounds very unnatural , especially on guitars .
I think the AN stuff tends to use the AD1865 , is that Burr-Brown ?
Although step-ups are very good in most respects, and help on the noise front, it has recently been shown to me how much they lose in the low bass and timing , presumably due to primary inductance issues . This thread :
audio-talk :: View topic - j-FET / Triode Phono front-end
Documents my experiments in building a jFET/triode cascode front-end for the phono, and the trials and tribulations of getting it to work . You will notice Joe Roberts and JC Morrison making an appearance in there, as some of JC's ideas ( see 'Lab JC' ) were being discussed.
At the end of the day, I haven't been able to get the jFET to work to my satisfaction yet, but it's shown me that there is a lot more to the low bass rendition than people using step-ups might realise. The change was NOT subtle !
However, the use of a 103 or 103R into step-ups is a sort of worst case really - the 103 being 40R and the 103R being 12R internal impedance. That tends to suggest the SPU should be the choice here , as it will have somewhere from 6R to 2R Zout , which should sort-out the bass issues to a large extent .
Anyway, that's enough of my ramblings ....
Phono interconnects require different consideration than interconnects between line level mainly because the source had a different characteristic. So from cartridge to the load, all elements and their effects need to be considered as an integrated performing circuit. Generally, balanced interconnects should perform better here.
Interfacing between audio equipment is quite a mess. When I discovered that unbalanced interconnects themselves dropped in impedance starting around 2KHz, I had designed them so that the impedance would be pretty much flat thinking that I solved a problem; only to find that some amplifiers design their inputs with ultrasonic filters that created the same problem as the interconnects. In the end, we just need to take care that all interfaces are tightly integrated.
Just browsed through the ESS DAC, I wonder which sound cards already have these 32 bit DACs in use. Certainly would like to get my hands on one.
Interfacing between audio equipment is quite a mess. When I discovered that unbalanced interconnects themselves dropped in impedance starting around 2KHz, I had designed them so that the impedance would be pretty much flat thinking that I solved a problem; only to find that some amplifiers design their inputs with ultrasonic filters that created the same problem as the interconnects. In the end, we just need to take care that all interfaces are tightly integrated.
Just browsed through the ESS DAC, I wonder which sound cards already have these 32 bit DACs in use. Certainly would like to get my hands on one.
Lynn:
I would be very interested if you actually get Audiotester and the 610 to work from Boot Camp. I was never able to get both to play nicely, and I suspect it had something to do with the FW implementation in Boot Camp. In the end, I ended up coughing up a new PC laptop with Win7. It isn't easy to find PC laptops with FW these days, and I spent a bundle.
Good luck and let me know if it is successful.
Hi Lynn
Very nice to see you back. I have been following this thread with interest, but have not been able to contribute until now.
First a quick question.
I have a Technics SP-15, a Technics SP-10 mk.II, a Linn LP-12 (old version) and a Lenco L-75. May I make a suggestion? Try to get hold of an old Lenco, and then have a look at the Lenco Heaven website. They have a lot written about how to tweak it to get the (very substantial) best out of it.
Enjoy,
Deon
Very nice to see you back. I have been following this thread with interest, but have not been able to contribute until now.
First a quick question.
I did a search on the Madisound website and I cannot find the drivers on there. Is the site just not updated yet? I would really like to build myself a pair of Ariels, but driver availability has stopped me dead in my tracks (until now).
Very, very nice! I have four turntables of my own.
I have a Technics SP-15, a Technics SP-10 mk.II, a Linn LP-12 (old version) and a Lenco L-75. May I make a suggestion? Try to get hold of an old Lenco, and then have a look at the Lenco Heaven website. They have a lot written about how to tweak it to get the (very substantial) best out of it.Now for another shameless plug for a friend- please investigate Lukasz Fikus' Lampizator DAC. There is a good review on the Stereomojo site. I think you might like it.
Enjoy,
Deon
Lynn, How would you use an ME2 as a center channel? I assume you would use it in a stand-up position just as the right and left channels. Wouldn't the offset tweeter cause directional issues?
Which leads me to ask: Would there be serious problems with an arrangement like this?
Which leads me to ask: Would there be serious problems with an arrangement like this?
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"Here's lookin' at you, kid"
An externally hosted image should be here but it was not working when we last tested it.
An externally hosted image should be here but it was not working when we last tested it.
FYI 103R is 19R really because I had measured it. Its printed 14R spec is off. Mine's two year old model. Same indication to another member when he checked. I use 220R load after that, and it was an improvement over 150R.
Thanks for the tip, much appreciated. Something to think about. There's plenty of information on the M-Audio forums. Here's an excerpt from a poster called "Tim":
---
I'm a new member and this is my first post. I've seen a number of complaints re: Pro Tools 8 and the ProFire 610 with iMacs on this and the DUC forum. I experienced similar problems and believe I've come across a solution.
For the record, I have a 3.06 core 2 duo iMac, 8gb Ram, OSX 10.6.4 (64 bit), Pro Tools M-Powered 8.04, and ProFire 610 firmware 1.01 and software version 2.2.4. I also record to a Glyph Gt 050Q external drive-1Tb.
I also got the "no M-Audio interface" messages on launch. In addition, there were times when the session would open, but when I hit "play", the session would freeze and the familiar "device not detected" message would pop up again. (I should note that with each scenario, I could avoid the problem by simply using the ProFire's external power supply--but this was not an acceptable solution IMHO).
So once I tried this set-up, I've had no problems whatsoever! Even phantom power to two Shure SM 81's works without a hitch.
1-Power down your iMac.
2-Use a firewire 800-800 cable to connect your external drive to the iMac's sole firewire port.
3-Then, use a firewire 800-400 cable to daisy-chain your ProFire 610 to the external drive. (The 800 side of the connector to the 800 of the drive of course, the 400 side to the ProFire) ***Do not use a 400-400 cable to connect the ProFire to the drive--this is when the problems start!!! I suspect that the bus power does not transfer from within the drive's 800 to 400 ports.
4-Power on the external drive, then the ProFire (flashing blue light), and then the iMac.
5-Once the system launches, the blue light on the ProFire will stop blinking and stay on.
6-Launch Pro-Tools, put the external power supply's away and have fun making music!!!
Any other minor glitches since have been easily rectified by turning off and then turning on the ProFire. This happened once or twice when I used the 610 as the audio output for iTunes and opened Pro Tools at the same time.
Also remember that firewire devices are not "hot swappable"---especially when carrying bus power.
---
Lynn here again. This post describes the native environment of the Mac, not using the Boot Camp partition to launch XP or Win7, which I'd need to do to launch ARTA or AudioTester. I could also try Parallels or VMWare, but somehow I doubt the high-speed Firewire interface would tolerate a virtualized PC. The thought of a Dell, Lenovo, or Panasonic Toughbook does not thrill me, since that would make it essentially a single-purpose PC. Will look further before jumping into this.
Returning to the topic of low-level phono interfaces, one of the unexpected aspects of a high-ratio step-up transformer is the transformation of impedances. The impedance ratio of a transformer is the square of the voltage, or turns ratio. This means if you're using a 1:10 step-up, and the phonostage input tube has 80 pF of Miller capacitance, it will look like 8000 pF at the transformer primary. Going further, a 1:20 step-up will make that same 80 pF look like 32000 pF! The assorted stray capacitances in the transformer only increase this figure.
That 80 pF figure is based on the somewhat sub-optimal choice of a 12AX7 as an input tube; a 6DJ8 would cut it about in half. To reduce input capacitance to the 2~5 pF region, you have to use a pentode - which has partition noise - or a cascode circuit. JFETs or bipolar transistors may have even more capacitance, unless they are used in a cascode circuit. Paralleling transistors, of course, multiplies both input capacitance and base currents.
A subtle disadvantage of a direct-coupled bipolar-transistor preamp circuit is the base current that flows through the cartridge, which magnetizes it over time. The same problem is seen in transistor tape decks, where the preamp electronics actually magnetize the heads. True, the input transistors can be AC-coupled, but this requires pretty large electrolytic coupling capacitors in the most sensitive node of the entire hifi system.
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