Preamp difference : if it's not the frequency, not the slew rate, and not the harmonics, what is it ?

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It does NOT contrary to popular belief eliminate skin effect, but does (in the correct application) markedly reduce copper losses and (in tuned circuits in the 100kHz to a few MHz) usefully raise the Q.

RF really isn't my area of strength but I did suspect the close proximity of the strands in an overall cable would mean they tend to behave wholistically rather than as individual wires, hence my half-question earlier.
 
Each DUT has its own non-linearity, which may well react differently to any artifacts in the aource A-D. These become like a multi-tone distortion stimulus, which different results among the DUTs. Best to have the cleanest possible input?
Very true - in a test of preamps this tests the whole system - the interface and the preamp with each other. No point in having the “greatest, punchiest preamp in the world” if it reacts badly with your interface. The DAW test is a good one if using internal bounces all on the same computer, which I did, as it truly tests the DAW. The same test system which I then used to compare interfaces, routes out through the interface and straight back in to compare different interfaces with each other from the same DAW with identical source material.
I was running a high tech pro-audio store at the time and we wanted our own opinions to escape the snake-oil merchants and glossy brochure claims - wanting to be able to give a proper answer to professional clientele who would come back to haunt you if you were wrong (or never come back at all which was worse!)
In the end it’s how things sound and work in the real world where they’re used and not on the test bench where the specs say “this must be the best”. I’ve worked in the recording industry since tape and before digital, transformer design was the holy grail along with tape speed and quality. Although tape rolled off at around 15 - 19 KHz depending on speed and tape type for multitracks, there was still plenty of audio presence above 20KHz if you looked and the filters were there to remove the bias frequencies and their artefacts, some above 100KHz bias freq - same game today - different gear, different problems and a different sound.
 
Sorry for the veer, but which DAWs did you test? I've long heard differences in DAWs, which many claim don't exist, but never really quantified it. One thing I've noticed is many don't dither properly.
 
Sorry for the veer, but which DAWs did you test? I've long heard differences in DAWs, which many claim don't exist, but never really quantified it. One thing I've noticed is many don't dither properly.
The DAW’s I tested were Cubase, ProTools, Logic Pro, Digital Performer and IIRC, Ableton. There was no dither, no samplerate or bitrate conversion - just a 44.1/16bit ripped straight from a commercially produced CD into each DAW, all faders at nominal 0dB so no volume changes, then a straight export or bounce (depending on the terminology of the DAW) straight to the internal HDD.
These bounces were all imported again into the same DAW - I think ProTools but I also tried import to others as well with the same net results.
Cubase and ProTools were sample identical - with one or the other phase flipped gave total total silence, all the others varied compared to these two and also to each other - changing levels did not change the differences only made them more, so it wasn’t a DAW volume thing.
As all the DAWS’s were on the same computer and only routing to the internal HDD there was no interface reaction or contributing factors to the test. The phase nulled summed track pairs were also re-exported and the resultant differential result imported into the benchmark DAW to see if there were any artefacts. Also of course we listened to the results with the gain cranked in the original comparison setup as well as the null track re-exports.
We did similar tests with varied interfaces as well comparing their D/A and A/D.
 
There was a lot of Blah at the time about various DAW’s having “superior audio engines” but this test seemed to confirm suspicions that some were more equal than others. You would have thought that an import of an already digital file would result in an identical file at the export end - but there is internal processing done to these files as well as export processing and therein lies the difference.
 
A straightforward 16/44 bounce not being correct is rather surprising. Maybe things are even more flawed than I thought.
Yeah - although this was done a little while ago, using Cubase and ProTools today and having also used Reaper from time to time I find they seem to give the best results. There seems to be an oddity in the pan law in Logic Pro by comparison. The busing, panner type selection and group routing in Cubase is great for mixdown as it’s like using a console in the box with extras. Also the zoom right down to individual samples is handy - being able to pencil redraw digital clips or copy and paste tiny chunks to fix them is really handy.
 
a very fast slew rate could potentially overload preamp circuitry

are you suggesting that by limiting the upper frequency response, we reduce distortion?

speakers can have intermodulation distortion

For sure! Loudspeakers tend to be the one element which adds considerable "colour" (aka distortion, aka frequency response that looks like the Himalayas, etc.) ... and yet being honest, while I sit here staring at my studio monitors which're truly excellent ... listening to them is sometimes like eating dinner without salt
 
are you suggesting that by limiting the upper frequency response, we reduce distortion?

Yes, both from eliminating RFI demodulation in the input stage, and by avoiding slew rate limiting in the second stage of the traditional three stage amplifier design. Slew rate distortion is also sometimes referred to as transient intermodulation distortion.
 
Yes, both from eliminating RFI demodulation in the input stage, and by avoiding slew rate limiting in the second stage of the traditional three stage amplifier design. Slew rate distortion is also sometimes referred to as transient intermodulation distortion.
It's kind of all the same thing. Rectification occurs because the circuit is just not fast enough for the negative feedback to match the input signal rate of change. TIM is just a cute name for the same thing. Slew rate limiting is like clipping in the rate of change domain.

I don't know if I have mentioned this yet this week(?), but Marshal Leach (RIP) published a topology where an audio path could be rise time limited and would be impossible to slew rate limit with a valid input signal.

JR
 
To answer OP's original question...

It's the biology of the circuit, the chemical composition.

the best "pre-amp" I ever had was a simple level control - a dual gang pot strung between the CD player's output and the power amp's input in that application. I'm no advocate of the ridiculous "passive preamps" sold by some people for extortionate sums of money but the short answer from me (for a change) is simplicity
 
Yes, agree !

Still fits my theory, as one element introduces less "chemical induced chaos" compared to many elements.

Did that simple dual gang pot have carbon tracks ?

I was just recently thinking about making a little switchbox with a stepped attenuator to connect my DAC's balanced outputs directly to the balanced inputs on my DIY Nelson Pass Amp, bypassing my mixer (hobbyist) homestudio setup.
For listening to music and watching movies, the idea was "this is going to sound better, and use less power from the socket"
 
Did that simple dual gang pot have carbon tracks ?

Might've been conductive plastic ... it would've been whatever I had lying around at the time.

In a properly decoupled system, it shouldn't matter because you only tend to get degradation (crackles, etc. when the pot's adjusted) if there's a DC offset running through the pot along with the signal or when there's significant deposition of dirt on the tracks ... smoking's an absolute killer for equipment (as well) because smoke + atmospheric moisture is acidic and eats switch contacts, faders, pots, etc.
 
PermO - I used a passive volume control that was on a very high quality dual rotary switch. Even though I didn't wire it to go to full signal strength (or even very close to it) it still took lots of (very high quality) resistors. In the end, I found the jumps between the click settings of the rotary switch were often just a bit louder or just a bit quieter than what I wanted to hear. So, I suggest that you us a stereo potentiometer to determine the resistor ratios that work for your tastes in your system before wiring up a complex switch system. Make sure you have at least one setting that is very, very quiet, but not completely off. I was surprised how often I wanted that. But in my circuit, the first "on" click was a bit too loud. Best of luck with it.
 

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