John Curl's Blowtorch preamplifier part II

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Undithered is different than dithered, AND requires averaging distortion into noise. IF you had music on a magnetic tape at -30Vu, and it was not a test tone, but a piece of music, would you say that it is digitized in measurable steps? Have you actually measured this? Compare that to a digital signal at -50dB, (approximately the same level) that is UNDITHERED. Dithering, in my opinion, is a bandaid, much like Tripath used oscillator modulation to HIDE the higher order harmonics. Prove me wrong, please.
 
IF you had music on a magnetic tape at -30Vu, and it was not a test tone, but a piece of music, would you say that it is digitized in measurable steps?

IF you had music on the output of a DAC and anti-imaging filter at -30Vu, and it was not a test tone, but a piece of music, would you say that it is digitized in measurable steps?

Compare that to a digital signal at -50dB, (approximately the same level) that is UNDITHERED.

Why would someone do something stupid like that? Add the same amount of noise as the tape. Or a whole lot less in most cases.
 
To jump ahead, PMA, another really interesting test to compare different recording characteristics would be to test an ANALOG SHIFT REGISTER using push pull pairs and cross coupled clocks. I built one of these, 36 years ago, and sounded pretty good, perhaps better than most digital, today. Here we sample, but do NOT break it into discrete bits. This seems, to me, to be more like analog tape recording, than a dithered digital 'equivalent'.
 
Undithered is different than dithered, AND requires averaging distortion into noise. IF you had music on a magnetic tape at -30Vu, and it was not a test tone, but a piece of music, would you say that it is digitized in measurable steps? Have you actually measured this? Compare that to a digital signal at -50dB, (approximately the same level) that is UNDITHERED. Dithering, in my opinion, is a bandaid, much like Tripath used oscillator modulation to HIDE the higher order harmonics. Prove me wrong, please.

what signal would you get out of the analog tape if you turned the AC bias off?

mag tape recording AC bias is just as much a "bandaid" as dither - one that made its use in "hi fidelilty" possible


some extol the virtues of SACD's DSD audio over PCM - but DSD is the extreme of noise shaped dither making 120 dB audio S/N possible from "one bit" digital
and DSD does "require averaging" - by a 6th order 50 KHz low pass filter
 
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Boy,

I overslept this morning and missed a private flight from my new found friend named "Andover", got back and worse yet my newly assembled interociter with electron sorter and holographic sound system was completely destroyed!

The sound system was really cool. It used the complex conjugate to derive the undistorted music image, then it used phased array inverse distorted ultrasonic transducers to reconstruct the original three dimensional sound field.

I'll have to rebuild it if I can find the parts.

In the mean time this is my thoughts on creating a very low level test signal to try and find the elusive yeti gap. As my scope does 128 averages every few ms it only needs to be 512 times better than that. I decided to start with a triangle wave since that is very easy to get constant level. Also to only look at zero crossings, distortion is not of great concern. As 1/f noise may be a problem I tried to avoid active components and resistors in the signal attenuator.

I show an extra gain stage on the output as the simulator scope really doesn't show nanovolts.
 

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Jee wiz, Nelson left an elephant in the livingroom and no one has noticed. Some audiophiles with $14,000 to spend on a cartridge/pre-amp don't need no stink'in RIAA. Maybe jcx has the deviation from ideal on the tip of his tongue but it's in the double digits I think. Just think of all the money saved on Teflon capacitors.
 
simple Bode asymptote aproximations: ~ 1:4 zero-pole "flat spot" is going to give ~ 12 dB difference from the expected pure 1st order slope of a uncorrected strain gage cart response
could be viewed as 12 dB mid/high cut or +/- 6 dB at low/high vs 1 kHz matched level
but in any case "celarly audible" by just about anyone in ABX comparison with a accurate RIAA played on the same system - pink noise anyone?

it could be excused on not doing level matched comparisons, by just walking into a demo room with unknown speakers/room frequency response - still I would be suprized that anyone with truely trained ears - even many "lesser trained" ears than recording/mastering engineers, techs wouldn't pick up on that much error even if they weren't sure where to place the blame

red single pole response - I assume even though strain gages give DC response any audio system won't like it so I kept the 50 Hz corner
green RIAA, the curves level matched at 1 KHz

looks like 38 degrees, 3.2 uS phase, time errors - where our hearing is pretty sensitive too
 

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Howard, your input is interesting, but misleading. Having designed a number of analog recorders, one thing stands out: AS you reduce level, distortion (3rd almost entirely) reduces right down to the noise floor, without dithering. No digital system that I know, can do this. Perhaps full 32 bit operation could approximate it. I don't know.

This is all entirely proveable with the level of digital equipment these days.

MSB Technology makes 32bit ADC's and DAC's with >140dB DR. This should
be enough to satisfy your low level distortion requirements. Do a reference
recording at 32bits then compare to bit reduced copies dithered down to 24
and 16 bits.

I'd be surprised if you could hear any difference at 24 bits and I think you
would be surprised at how little difference the 16 bit version would be.

T
 
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Jee wiz, Nelson left an elephant in the livingroom and no one has noticed. Some audiophiles with $14,000 to spend on a cartridge/pre-amp don't need no stink'in RIAA. Maybe jcx has the deviation from ideal on the tip of his tongue but it's in the double digits I think. Just think of all the money saved on Teflon capacitors.

Scott, do you mean this:
http://www.diyaudio.com/forums/analog-line-level/146693-john-curls-blowtorch-preamplifier-part-ii-971.html#post2464674
... The single pole integration does result
in some deviation, but no customers seem to have minded,
and it did sound good.


Which in turn refers to:
http://www.diyaudio.com/forums/analog-line-level/146693-john-curls-blowtorch-preamplifier-part-ii-969.html#post2464203
Sound-Smith aduiphoolish bable ignores basic linear systems theory

they blow their credibility for any serious engineer with the claim:

"The Strain Gauge cartridge is a displacement device, producing an output that is dependent and directly proportional to the amount of displacement of the stylus, NOT the velocity, like magnetic cartridges. It therefore inherently plays the RIAA encoded groove correctly. If one inspects the RIAA EQ, one will discover that there are discrepancies that occur where a displacement type cartridge deviates from a perfectly flat playback of the recorded RIAA grooce, and therefore will not produce a perfectly flat response.

Attempts made in the distant past to absolutely correct Strain Gauge cartridges for any amplitude anomaly have required equalization circuits, which not only add circuitry, but also introduce phase or time shifts. It is the Soundsmith’s belief that human hearing is much more forgiving of amplitude errors than time errors, so we have made minimal efforts to correct for any amplitude deviation from absolute flatness.
" Strain Gauge details

if they do get a good built-in "compensation" for for the 1st order slope between velocity and displacement then they still have the 500/2122 Hz shelf to EQ - not doing so results in More "phase or time shifts" since the EQ preemphasis and correction is "minimum phase" - certainly for mid band audio frequencies - any amplitude error is exactly related to a phase error by Bode's Integral of Phase

?

Regards
George
 
This is all entirely proveable with the level of digital equipment these days.

MSB Technology makes 32bit ADC's and DAC's with >140dB DR. This should
be enough to satisfy your low level distortion requirements. Do a reference
recording at 32bits then compare to bit reduced copies dithered down to 24
and 16 bits.

I'd be surprised if you could hear any difference at 24 bits and I think you
would be surprised at how little difference the 16 bit version would be.

T

32 Bits! Wow! WOW! WOW!

150 ohm resistor at room temperature 20khz bandwidth noise, is .22uv, 32 bits is times 429,496,729 or 944.9 volts from the noise floor to full range output!

Can a microphone do this? What happens to my amplifier input if the D/A clips? From 0 db to 193 db! Gee my speaker can almost reach total atmospheric modulation! Do I need two planets for stereo?

140 db is 140 db or 10,000,000 to 1. or 23.25 bits and that .25 bit is a real B!

You are right you would not hear a difference at 24 bits!
 
Given that velocity is the differential of displacement, and displacement is the integral of velocity, it is unclear to me what is being claimed for strain gauge or other displacement cartridges. A pure displacement cartridge still has to track the spiral groove, so it needs a mechanical high-pass filter or something similar. As the quote says, you can't separate amplitude from phase (except by using nonminimal phase filters). The mid-band kink (318/75us) still has to be there.

Have I missed something?
 
I pre-emphasized 2mV, 100 Hz + 1 mV, 5 KHz square waves and put them thru the 1st order 50 Hz LP and full RIAA eq

with 1 KHz matched EQ levels for minimum FR error you can see red single pole "uncorrected strain gage cart" response is wrong at both low and high frequencies

but what's +/- 6 dB error as long as there's a good audophile story attached?
 

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To arive at the optimum position a mono record could be played and a scope fed with a left - right channel signal in antiphase could show the position of minimal crosstalk.

NO, NO, NO, NO, NO!!!

Sorry Joachim, nothing personal, but this is one of the most widely perpetuated myths concerning vinyl.

Your system will only equalize the output levels of the two channels. But that is NOT what azimuth adjustments are for. The output level only varies slightly with azimuth changes. But the crosstalk is extremely sensitive to changes in azimuth.

The only way to set the azimuth properly is with a special test record that has a signal cut only in one channel and then in the other. Then you need a way to measure the output of the cartridge so that the crosstalk can actually be measured. This can be very difficult with MC cartridges.

Thirty years ago all good dealers in the US had a little box made by Shure that had a meter to show the output level. You would calibrate the gain to read 0 dB on the active channel and then switch to the other channel to read the crosstalk level. Then you had to repeat for the other direction. In those days there were no tonearms that allowed for azimuth adjustment on the fly so it took about an hour just to set the azimuth. And that is if you had all of the proper equipment.

Today I think that Feickert makes a computerized version to do a similar thing. But if it doesn't come with a special test record, it cannot work properly.

Sorry this reply is so late. If it is duplicating someone else's post, I am doubly apologetic.
 
This was the triumph of the Dolby HX system. When the HX detector frequency response curves were carefully adjusted for a specific tape type and static bias point, the sonic result was remarkably superior to a fixed bias approach due to stabilization of this bias bubble. I spent hundreds of hours with Dolby engineers perfecting this system, and the imaging and clarity of pure tones embedded in music, with removal of FM sidebands from the sound was amazing.

Howard Hoyt

WOW!

Thanks for a great post!

For those of you too young to remember, what Mr. (Dr.?) Hoyt is talking about was the fact that there is an optimal bias point for each formulation of recording tape. Setting the bias was SOP before all pro recording sessions, and if you bought a consumer-grade recorder the bias was never quite right. They would always just set it to be a certain voltage on the production line, but variations in the head would mean that it was never quite right.

Paying a tech $100 or so to "align" the entire machine (electrical and mechanical) paid big sonic dividends. But when you had the machine aligned, you would always have it set for the specific brand and formula of tape that you planned to use.

People knew that the high-frequency content in the audio signal acted somewhat like a bias signal and "interfered" with the proper bias level. So in the twilight of analog recording, Dolby developed a variable bias system that took into account the audio signal to keep the overall bias much more linear and accurate.

When the system was introduced, all of the magazines had articles explaining it and measurements showing the improvement it made. IIRC, B&O were the first to use it, but I could be mistaken. Unfortunately within a few years, Dolby HX was lost to the digital onslaught. I can't remember if any of the pro machines ever used it or not. It wasn't nearly the problem at 30 ips as it was at 1-7/8 ips! (I'm sure that Mr. Hoyt will correct me if I am wrong about that.)

Thanks again for the great post!
 
Stylus Cantilevers

Joachim's post on the diamond coating of a boron cantilever is very interesting. There are a lot of unknowns here. My first exposure to this was when Sumiko had a line of three cartridges called Alchemist Talisman or something like that. One had an aluminum cantilever, one had a sapphire cantilever, and one had a boron cantilever. These were allegedly the only differences.

The sapphire was a huge step up over the aluminum. Much more clarity and resolution. Unfortunately it also had a clear peak at the high frequencies. Whether this was a resonance in the cantilever or simply a more accurate translation of the resonance of the mass of the stylus in the compliance of the vinyl groove, I have no idea. But it was not pleasant to listen to.

The boron was perfect. Beautiful, musical, clear, resolved, everything. Since then, all of my favorite cartridges featured boron cantilevers. So now we have two posts pulling us in opposite directions.

One says that a very expensive and very esoteric cartridge with an aluminum cantilever is the best he has ever heard. Personally I have a hard time believing that replacing the cantilever with a boron one wouldn't make a further improvement.

The other post (by Joachim) talks about stiffening the boron cantilever by coating with a layer of amorphous diamond crystals. The obvious question is if this is so great, why not just use a solid diamond cantilever as some companies have done before?

I know that the sapphire cantilever did not sound as good as the boron. Was it because the sapphire was not stiff enough? Would a diamond cantilever be the ultimate? Or is the mass of the diamond too great, resulting in a lower resonance frequency between the effective tip mass and the compliance of the vinyl.

Remember, there are at least three separate resonances in the modern MC cartridge:

a) Effective tip mass and the compliance of the vinyl groove

b) Bending mode resonances in the cantilever (other modes will be too high in frequency to matter)

c) There is usually a "tie-wire" holding the cantilever in place. It is typically a metal wire and the tension is individually adjusted on each cartridge to produce the flattest response. But this is basically just like a very short violin string and will have many resonance modes.

I would guess that much of the art of MC cartridge design is balancing these three resonances. If they are not ALL moved well outside the audio band, there will be audible colorations. That is when even more "black art" is involved, changing the body material and so forth in attempts to mask or balance these audible resonances.

I don't know. But I bet that some of the people on this forum have some pretty good ideas.
 
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