Why do some audiophiles spend thousands of dollars on a DAC? Are they searching for a "sound signature" they like, or just greater "accuracy"?

[Satanica]

6 minutes ago, andyr said:

Whilst, I agree with you, S ... I'm not sure MLXXX can comprehend that two components can measure the same (at least, as far as the measurements we know how to do, anyway) - yet one can sound more 'musical' (= 'pleasing', to me   ).

 

To be fair Andy, I'm not sure I can either.

[aussievintage]

9 minutes ago, andyr said:

18 minutes ago, Satanica said:

 

A more "musical" sound may or may not imply that it was the deliberate intention of the vendor but more importantly it does not result in a measurable change in frequency response, noise reduction, distortion reduction etc.

 

 

Whilst, I agree with you, S ... I'm not sure MLXXX can comprehend that two components can measure the same (at least, as far as the measurements we know how to do, anyway) - yet one can sound more 'musical' (= 'pleasing', to me   ).

 

 

The trouble is that my mind just screams out wanting to know wtf the difference is, that it won't show up in the usual measurements.    So there's something else different? Great, then I want to know how to measure it - that sort of puzzle just niggles away at me, like the name of a song that you just can't recall.    

 

I won't be happy until we find out.   It's like that saying, if there's no pictures, then it didn't happen.

[Guest DrSK]

1 hour ago, aussievintage said:

 

 

The trouble is that my mind just screams out wanting to know wtf the difference is, that it won't show up in the usual measurements.    So there's something else different? Great, then I want to know how to measure it - that sort of puzzle just niggles away at me, like the name of a song that you just can't recall.    

 

I won't be happy until we find out.   It's like that saying, if there's no pictures, then it didn't happen.

I firmly believe the standard spec sheets are measuring the wrong thing.

 

But they are what market demands, based on years of historical measurements back when tonal frequency sweeps and tonal distortion measurements were all that could be measured. I don't think measurements for audio have moved on at all and are deficient, given the real differences in equipment with similar spec sheets. 

[andyr]

1 hour ago, Satanica said:

 

To be fair Andy, I'm not sure I can either.

 

 

I'm happy to choose components by how they sound - ie. the kick I get from listening to them ... and don't really gaf what they measure like.  

 

1 hour ago, aussievintage said:

 

The trouble is that my mind just screams out wanting to know wtf the difference is, that it won't show up in the usual measurements.    So there's something else different? Great, then I want to know how to measure it - that sort of puzzle just niggles away at me, like the name of a song that you just can't recall.    

 

 

I think the key there, av, is "the usual measurements "!  IOW, the ones we know how to do.  

 

OK ... let me put a scenario to you (involving 2 amps) and perhaps you would tell me what you would measure, in order to show why amp #2 did what it did.    (The 5 people at this listening session all heard the same thing.)

 

The scenario is as follows:

• you are listening to amp #1 and then amp #2.
• these are driving the same pair of spkrs and have the same preamp & CDP in front of them.  And you are listening to the same CD - involving a female singer - with both amps.
• IOW - the only change is ... amp #1 is taken out and replaced with amp #2.
• with amp #1, the singer is in the same plane as the speakers.  So yes, there is depth behind the spkrs - but no sound stage in front of them.
• with amp #2 - the singer jumps forward of the spkrs about a metre.  And there is also depth.

Both amps were ss:

• amp #1 was a well-known US brand with several hundred watts on tap
• amp #2 was a single-ended amp with 25w.

 

Andy

 

[andyr]

10 minutes ago, DrSK said:

I firmly believe the standard spec sheets are measuring the wrong thing.

 

But they are what market demands, based on years of historical measurements back when tonal frequency sweeps and tonal distortion measurements were all that could be measured. I don't think measurements for audio have moved on at all and are deficient, given the real differences in equipment with similar spec sheets. 

 

Hear hear!  

 

(We can sit together on the "excommunicated" bench!  )

 

Andy

 

[aussievintage]

2 hours ago, andyr said:

I think the key there, av, is "the usual measurements "!  IOW, the ones we know how to do.  

 

 

Bugger I lost a fairly long post about this just a minute ago.

 

To summarise, I think terminology like "musical" is an aggregational term for a whole bunch of small differences that we can already measure.  Separately, each small difference is not seen as pertinent.    We just haven't put the clues together to recognise what we need to bring into conjunction to make it happen.   

 

Sometimes, accidentally, these things just happen in a system, and it causes the listener to say it sounds more "musical" .  It doesn't help that it is probably different for each listener (but not always).  When people say the specs are similar, that also actually means there are small differences.     What are they?  Hmmm, don't know.  Some usual suspects may be part of it - harmonic content , amount and type, for one example.  That nearly flat frequency response, may still be bumpy enough to subtly change the flavour maybe?   I'd go for, measure every damn thing you can while doing subjective, but controlled testing, and look for a statistical correlation between any and all.

 

Edited November 13, 2020 by aussievintage

[tripitaka]

On 02/11/2020 at 8:24 AM, aussievintage said:

 

Already doing it on my raspberry Pi.   Mainly using filters and a "tube warmth" effect.  I'd welcome suggestions for other effects to try  

 

 

 

Which program, please?

 

....this should save me from a pretty expensive experiment

[aussievintage]

2 hours ago, tripitaka said:

 

Which program, please?

 

....this should save me from a pretty expensive experiment

 

This little project, while predominantly a phono preamp, also has these features added.  It uses a "Puredata"  with LADSPA plugins (for things like the tube warmth)

 

eg.

 

 

 

Edited November 13, 2020 by aussievintage

[Stereophilus]

16 minutes ago, andyr said:

 

I'm happy to choose components by how they sound - ie. the kick I get from listening to them ... and don't really gaf what they measure like.  

 

 

I think the key there, av, is "the usual measurements "!  IOW, the ones we know how to do.  

 

OK ... let me put a scenario to you (involving 2 amps) and perhaps you would tell me what you would measure, in order to show why amp #2 did what it did.    (The 5 people at this listening session all heard the same thing.)

 

The scenario is as follows:

• you are listening to amp #1 and then amp #2.
• these are driving the same pair of spkrs and have the same preamp & CDP in front of them.  And you are listening to the same CD - involving a female singer - with both amps.
• IOW - the only change is ... amp #1 is taken out and replaced with amp #2.
• with amp #1, the singer is in the same plane as the speakers.  So yes, there is depth behind the spkrs - but no sound stage in front of them.
• with amp #2 - the singer jumps forward of the spkrs about a metre.  And there is also depth.

Both amps were ss:

• amp #1 was a well-known US brand with several hundred watts on tap
• amp #2 was a single-ended amp with 25w.

 

Andy

 

I think the known measurements for audio gear are absolutely vital for the initial engineering and design.  However listening will often pick up qualities that can not always be fully explained by the measurements.  Both listening and measurement are essential to good audio equipment design IMO.

 

The conundrum of what is being missed in our measurements is akin to the “hidden variable theory” in quantum mechanics.

https://www.google.com.au/url?sa=t&rct=j&q=&esrc=s&source=video&cd=&ved=2ahUKEwij8qXMkf_sAhXI4zgGHfgKDas4ChC3AjAAegQIAhAC&url=https%3A%2F%2Fwww.youtube.com%2Fwatch%3Fv%3De0GhlCzLmN4&usg=AOvVaw29VAfufCzMZWS2O9sjmLp6

 

The answers won’t be found in believing it’s all magic, and likewise, the answers won’t be found by ignoring clear observational evidence.  

 

[blakey72]

Well I am about to genuinely start looking for an external dac to stream through and perhaps play hi-res downloaded files through. I've been reading this thread since it started and have been thinking about what I think I'm after. I guess I'm after a few things:

 

1. I want the device to be able to do certain things - Decode MQA, play resolution high enough to ensure it won't be outdated too soon. 

2. Hopefully improve sound quality over what I have at present.

3. Have a sound that I love and excites/impresses me when I hear it. 

 

What I am not too concerned about:

1. What price it is (within reason of course). 

2. What other people think (bragging rights).

3. Accuracy - this is a strange one. I like my music to be accurate but not at the cost of not liking the sound. I used to think accuracy was the be-all and end-all, I don't think as much in that way now. I prefer to just like what I hear, hard to explain. I guess I'm just trying to enjoy the music more however it comes and worry less about how 'my system' sounds. 

[aussievintage]

2 hours ago, andyr said:

•  
• with amp #1, the singer is in the same plane as the speakers.  So yes, there is depth behind the spkrs - but no sound stage in front of them.
• with amp #2 - the singer jumps forward of the spkrs about a metre.  And there is also depth.

 

 

Just was thinking about this while watching Netflix in front of my relatively good soundbar.     Positioning outside of the plane  or line of the sound sources has got to be brought about by relative phasing fooling the ear.  The soundbar does this exceptionally well.  This perceived improvement in the two amp comparison may be just a form of distortion that is different between the 2 amplifiers.   I can't discount that the lack of depth may be the true nature of the recording, and the improved staging is just an effect.

[aussievintage]

46 minutes ago, blakey72 said:

I used to think accuracy was the be-all and end-all

 

 

So did I until I had a software control that dialed in artificial harmonics, and there was no doubt I preferred the sound with the added distortion.  Dare I say it - it was much more "musical"

[andyr]

11 minutes ago, aussievintage said:

 

I can't discount that the lack of depth may be the true nature of the recording, and the improved staging is just an effect.

 

 

Aah but that's just you, av!

 

I can't discount that the forwardness was the mic capturing the recording exactly - ie. that was the true nature of the recording - and the lack of forwardness in amp #1 is simply due to a lack in its design.  

 

13 minutes ago, aussievintage said:

 

So did I until I had a software control that dialed in artificial harmonics, and there was no doubt I preferred the sound with the added distortion.  Dare I say it - it was much more "musical"

 

 

"Musical" - you mean with added H2!  

 

(That's why tubes are so loved.)

 

Andy

[aussievintage]

Just now, andyr said:

Aah but that's just you, av!

Absolutely true.

 

Just now, andyr said:

I can't discount that the forwardness was the mic capturing the recording exactly - ie. that was the true nature of the recording - and the lack of forwardness in amp #1 is simply due to a lack in its design.  

 

 

What was the recording?  The typical studio recording done with separate mics and separate tracks then mixed together is all artificial anyway.    Now if it was a live recording done the good old way with 2 or 3 microphones in front of the performance...

 

 

2 minutes ago, andyr said:

"Musical" - you mean with added H2!  

 

(That's why tubes are so loved.)

 

 

Yep, that's why I don't use the software on my main system.  It's also behind  the magic often spoken about when listening to SET amps.  I know it gets up some people's noses, but a my little 2A3 stlye SET through those efficient Osborns,   just sings musically all day.

[Guest DrSK]

4 hours ago, andyr said:

 

I'm happy to choose components by how they sound - ie. the kick I get from listening to them ... and don't really gaf what they measure like.  

 

 

I think the key there, av, is "the usual measurements "!  IOW, the ones we know how to do.  

 

OK ... let me put a scenario to you (involving 2 amps) and perhaps you would tell me what you would measure, in order to show why amp #2 did what it did.    (The 5 people at this listening session all heard the same thing.)

 

The scenario is as follows:

• you are listening to amp #1 and then amp #2.
• these are driving the same pair of spkrs and have the same preamp & CDP in front of them.  And you are listening to the same CD - involving a female singer - with both amps.
• IOW - the only change is ... amp #1 is taken out and replaced with amp #2.
• with amp #1, the singer is in the same plane as the speakers.  So yes, there is depth behind the spkrs - but no sound stage in front of them.
• with amp #2 - the singer jumps forward of the spkrs about a metre.  And there is also depth.

Both amps were ss:

• amp #1 was a well-known US brand with several hundred watts on tap
• amp #2 was a single-ended amp with 25w.

 

Andy

 

 

I'd start by measuring the signal at the listener location using a head and torso simulator. Or at least a binaural dual channel microphone setup. Run mics with range around 30kHz so they can't affect anything. 

 

Repeat with both amps and overlay the signals and compare.

 

Then repeat but with mics in the speaker direct field.

 

I'd oversample the source if digital by at least 10 times as nyquist limits don't cut it for capturing waveform details. For analogue source I'd still run 10 times CD sample rate. 

 

Gut feeling is I'd expect to see differences in attack for the vocals and introduced transients.

 

Higher attack, decay and cleaner signals bring sound forward. Some DAC chips can alter this in the filtering to change how up front the sound is or introduce some transients to change the power of the sound. 

 

I also watched a live HD stream of a band at a Sydney venue which is an emerging thing with covid. It was interesting as I could see as some of the drum kit was very close mic'd (eg the brass) whereas the skins weren't. The difference in attack was very apparent and the close mic stuff popped out in front of my speakers whereas the skins were 2m behind my speakers. I spoke to the guys that did the mic setup and mixing, they couldn't hear this but were using some not very good headphones. All they could hear was stereo. My hifi got the stereo but also depth that was distracting as it didn't match the physical location of the instruments. 

 

I need to get some recordings of a drummer soon for work. It will be mono but will play with mic placement and see how distance to the drums changes how forward the sound is on playback. Might do the same just talking too. 

 

This is all stuff I'd love to measure and relates to my profession. Just need time, am currently lucky in that I have clients already booking me up until June next year. 

 

 

Edited November 13, 2020 by DrSK

[MLXXX]

6 hours ago, Satanica said:

A more "musical" sound may or may not imply that it was the deliberate intention of the vendor but more importantly it does not result is a measurable change in frequency response, noise reduction, distortion reduction etc.

You have stated how you are using the word "musical", in the current context. 

 

It does strike me as a little odd that a DAC could sound different but be exactly the same as regards measured " frequency response, noise reduction, distortion reduction etc.".  I'd note that DACs can vary a little in their frequency response as they get towards their upper (Nyquist) limits, and that variation is more likely to have an audible effect if a DAC is fed a relatively low sample rate stream such as the 44.1kHz from a standard CD. 

 

6 hours ago, Satanica said:

No I'm not talking about "enhanced" sound because that implies it was a deliberate and more importantly a measurable change in frequency response.

Thus your recommendation of a sound processor.

Actually I was not thinking of a sound processor as just being used to change the frequency response, although that is the classic use for them. For example, a sound processor could change the stereo imaging, expand or compress the dynamic range, or add reverberation. 

Edited November 13, 2020 by MLXXX

[MLXXX]

On 07/11/2020 at 11:37 PM, kukynas said:

if you can compile the test file, tell me which soft you'd like to use for playback/recording I can run it through and send you the output files

Actually, kukynas, I think I've come up with a better test file now than the one I suggested to you earlier.  It starts with a short burst of 11025Hz sine wave at -6dB, and ends with a longer burst of 1000Hz sine wave, also at -6dB. In between is beautiful music!  I downloaded the music from the excellent HiRes download test bench webpage maintained by 2L the Nordic Sound.  It is the 9th item on the webpage: Magne Amdahl: Astrognosia - Aquarius, Norwegian Radio Orchestra, which runs for a minute and a half.  I selected the "original CD 16BIT/44kHz" version.

 

Using my own equipment, a Behringer FCA1616 audio interface,  as both a DAC and an ADC, I was nowhere near achieving bit-for-bit accuracy. Although the "round trip" (DAC to analogue cable to ADC) frequency response was well maintained at low frequencies, it tapered off significantly from about 10kHz.  This was using a sample rate of 44.1kHz, the same rate as the music file. 

 

Via Google Drive I can share with the forum the re-recordings I've made. The files are:

 

1. The original as downloaded,  but with opening and closing tones added, and at 24 bits 44.1kHz rather than the original 16 bits 44.1kHz format.
2. The above after a single "round trip" of being played back with a DAC and simultaneously re-recorded with an ADC.
3. A later generation copy, involving X round trips from the original downloaded file.  Forum members are welcome to guess how many round trips were involved.  Does X equal 2, 3, even more?
4. An even later generation copy, involving exactly twice as many round trips.  

 

The files are labelled and it is not necessary to guess which is which. Here they are: The 4 test files     The only unknown forum members are being asked to guess is how many passes through the audio interface were involved for file 3 [being half as many as for file 4].   

 

This exercise may serve to give a broad idea of how much sound degradation to expect when doing multiple conversions between digital and analogue with ad hoc devices. I suspect some people may be surprised at how well the sound holds up. On the other hand, degradation is audible even with a single round trip. (By the way, the first file is inverted in phase relative the other three.)

 

I should mention that the line output DACs in the Behringer interface are not a focal point of its design. It is more important as a recording device.  I should also mention that the front panel TRS inputs, used as line inputs,  involve a pre-amplifier prior to the ADC.   In the photos below you can see where the analogue cables connected at the rear of the FCA1616 (to its outputs "5" and "6"), and to the front of the unit (to it's inputs "3" and "4"). 

 

The DAW software used was Tracktion Waveform Pro 11. The gain was set such that each re-recording lost about 0.2B in each channel, though there was a little bit of gain fluctuation during the re-recording session probably due to rising temperature. In later editing, the 1kHz tone at the end of the file was used as a reference to restore the recorded files to the original amplitude for the 1kHz tone. The audio file editor Audacity was able to perform this role to the nearest thousandth of a dB!  It is amazing what precision is available when manipulating audio files!  It was also necessary to re-align the files to the nearest sample at 44.1kHz, because of latency delay.

 

 

 

 

[Ittaku]

4 hours ago, blakey72 said:

Well I am about to genuinely start looking for an external dac to stream through and perhaps play hi-res downloaded files through. I've been reading this thread since it started and have been thinking about what I think I'm after. I guess I'm after a few things:

 

1. I want the device to be able to do certain things - Decode MQA, play resolution high enough to ensure it won't be outdated too soon. 

2. Hopefully improve sound quality over what I have at present.

3. Have a sound that I love and excites/impresses me when I hear it. 

 

What I am not too concerned about:

1. What price it is (within reason of course). 

2. What other people think (bragging rights).

3. Accuracy - this is a strange one. I like my music to be accurate but not at the cost of not liking the sound. I used to think accuracy was the be-all and end-all, I don't think as much in that way now. I prefer to just like what I hear, hard to explain. I guess I'm just trying to enjoy the music more however it comes and worry less about how 'my system' sounds. 

 

Look to MSB DACs, though the price "within reason" may fail there by your standards. It's hard to know what reason is to you.

[MLXXX]

15 hours ago, andyr said:

• with amp #1, the singer is in the same plane as the speakers.  So yes, there is depth behind the spkrs - but no sound stage in front of them.
• with amp #2 - the singer jumps forward of the spkrs about a metre.  And there is also depth.

Both amps were ss:

• amp #1 was a well-known US brand with several hundred watts on tap
• amp #2 was a single-ended amp with 25w.

 

The above scenario relates to the thread topic in that it is an example of where a noticeable audible effect may be due to non-standard audio device behaviour.  Both amps can't be right!  Do we choose a more accurate "standard" amplifier or a less accurate "exotic" one?  If the exotic one, will it be pleasing all the time, or only in special circumstances?  Will it sometimes be less pleasing than the standard amplifier?

 

The above scenario also  relates to the question of using sound processors, as one of their possible functions is to create interesting spatial effects. Doing something with sound that is common to both channels (such as a single microphone used for a vocalist and panned equally to Left and Right) so as to give an artificial spatial effect has been an available feature for a very long time, even before DSP took off in consumer audio equipment. 

 

(I'd be inclined to check the performance of amp #2 first. It is hard to avoid phase shifts with a valve operating in Class A driving an audio output transformer.  On the other hand if amp #1 uses DSP, there may be an issue with the speaker distance setting, a setting that can introduce a timing delay.)

Edited November 14, 2020 by MLXXX

[Guest DrSK]

4 hours ago, MLXXX said:

 

The above scenario relates to the thread topic in that it is an example of where a noticeable audible effect may be due to non-standard audio device behaviour.  Both amps can't be right!  Do we choose a more accurate "standard" amplifier or a less accurate "exotic" one?  If the exotic one, will it be pleasing all the time, or only in special circumstances?  Will it sometimes be less pleasing than the standard amplifier?

 

The above scenario also  relates to the question of using sound processors as one of their possible functions is to create interesting spatial effects. Doing something with sound that is common to both channels (such as a single microphone used for a vocalist and panned equally to Left and Right) so as to give an artificial spatial effect has been an available feature for a very long time, even before DSP took off in consumer audio equipment. 

Hate this! Pay good money to get good imaging, then the vocals get split in frequency content to left and right to widen the image so there is no distinct image. 

 

Quote

(I'd be inclined to check the performance of amp #2 first. It is hard to avoid phase shifts with a valve operating in Class A driving an audio output transformer.  On the other hand if amp #1 uses DSP, there may be an issue with the speaker distance setting, a setting that can introduce a timing delay.)

Interested in how phase shift brings the sound forward?  Would have thought if anything it weakens the image as phase shift will be frequency dependent. Compared to actual sounds when created and propagated where there is a change in frequency content and decay of peaks with distance. Eg decay of plosives etc. Or in many instances, the sound gets less clean with distance as direct to reflected time delay and level ratios change 

Edited November 13, 2020 by DrSK

[MLXXX]

2 hours ago, DrSK said:

Hate this! Pay good money to get good imaging, then the vocals get split in frequency content to left and right to widen the image so there is no distinct image. 

I have never been a subscriber to interfering with a recording spatially. If it's in mono let it stay that way. If it's in stereo I'm not interested in simulated surround. (In a video context, on-the-fly simulation of 3D from 2D program material is more distracting to me than pleasing, and of course it is artificial and potentially highly false and misleading.  Painstaking frame by frame simulation of 3D is another matter: it has the potential to produce fine results.) Looking much further ahead in time, it's possible spatial enhancement processing for music will reach a very advanced level using artificial intelligence, and if I'm still alive then, I might need to reconsider my "purist" position.

 

In the realm of DACs, at this point in time I would be in the camp that looks for as close as possible to a "neutral" and "accurate" analogue output. As has been mentioned earlier in this thread, if a selection of very high end DACs sound noticeably different to each other we have to ask ourselves why.  Let me ask a specific question:  

 

Are  the reported differences in sound quality produced by high end DACs more obvious to the ear when the DAC is being used for the playing of a standard CD (a source with a sample rate of 44.1kHz), as distinct from using the DAC for material in a format with a sample rate of 48kHz or higher?

[aussievintage]

7 hours ago, MLXXX said:

If the exotic one, will it be pleasing all the time, or only in special circumstances?

 

I have read the suggestion before, that (in my case) a SET amplifier is only good for certain types of music.   I no longer believe that the problem has anything to do with the harmonics.  When a SET amp struggles with a certain recording, it is for other reasons.  Chiefly, lack of power, especially in the bass regions.  I now drive the low end (125 Hz down) with a class D amp, and let the SET do all the rest.    I play absolutely ALL types of music, and they all sound good IMHO  

 

As for the harmonics sounding good, this is also true for any type of music.  I have tried it, adding them to the input of a straightforward class AB solid state amp,  on piano solo, female vocal, rock band, big swing band, full orchestras.   It is always an improvement when added subtly. 

[Ittaku]

29 minutes ago, MLXXX said:

Are  the reported differences in sound quality produced by high end DACs more obvious to the ear when the DAC is being used for the playing of a standard CD (a source with a sample rate of 44.1kHz), as distinct from using the DAC for material in a format with a sample rate of 48kHz or higher?

This one is more of a "depends". Some DACs seem to improve more with 44k material than highres, whilst others are better across the board. Better filtering is enough to explain the former group, whilst the latter group... well dunno. Could be any number of things, but likely it's all the "things".

Edited November 14, 2020 by Ittaku

[andyr]

7 hours ago, MLXXX said:

 

The above scenario relates to the thread topic in that it is an example of where a noticeable audible effect may be due to non-standard audio device behaviour.  Both amps can't be right!

 

Your comment that "Both amps can't be right! " is completely bizarre to my way of thinking.

 

AFAIAC, both amps were exhibiting "standard audio device behaviour " - but the sound they delivered had different artefacts.

 

7 hours ago, MLXXX said:

Do we choose a more accurate "standard" amplifier or a less accurate "exotic" one?  If the exotic one, will it be pleasing all the time, or only in special circumstances?  Will it sometimes be less pleasing than the standard amplifier?

 

Good points!    That's why A-B instant comparisons are useless - you generally need to have an extended listen, to decide which component you find more pleasing.

 

7 hours ago, MLXXX said:

(I'd be inclined to check the performance of amp #2 first. It is hard to avoid phase shifts with a valve operating in Class A driving an audio output transformer.  On the other hand if amp #1 uses DSP, there may be an issue with the speaker distance setting, a setting that can introduce a timing delay.)

 

Mmmm. your comprehension of what ha beens written seems to be slanted towards your own biases?  

 

I said both amps were ss - in spite of amp #2 being a SE design.  No tubes were involved.

 

And amp #1 was a Bryston (from about 20 years ago). It did not incorporate any DSP.

 

Andy

 

[aussievintage]

5 minutes ago, andyr said:

Good points!    That's why A-B instant comparisons are useless - you generally need to have an extended listen, to decide which component you find more pleasing.

 

They are NOT useless.  Maybe not helpful for this particular test, but don't throw the baby out with the bathwater  

 

6 minutes ago, andyr said:

I said both amps were ss - in spite of amp #2 being a SE design.  No tubes were involved.

 

A SE FET would sound similar to a SE valve w.r.t. harmonics.  What was this amp?

 

8 minutes ago, andyr said:

Your comment that "Both amps can't be right! " is completely bizarre to my way of thinking.

 

AFAIAC, both amps were exhibiting "standard audio device behaviour " - but the sound they delivered had different artefacts.

 

Can one not still be "right" ?