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“Human ears are better than measurements!” / “Humans can hear things we can’t even measure!”

Who says those adjectives are right? ASA didn't, right?

Even your representation of what "reviewers" says is without foundation. When they mean "less warm," they say the sound is bright. But go ahead and show us how reviewers universally mean that.

Back to reality, frequency response errors in mid-range massively change its characteristic in the way they push that range/vocals back or bring it forward. Anything else is made up nonsense.
Yes, frequency response magnitude changes midrange presence. So does harmonic distortion spectrum, even at levels below typical audibility thresholds in isolated sine tests. So does damping factor and its interaction with speaker impedance. So does temporal behavior and group delay. You regularly measure most of these.

The claim wasn't that reviewers are right about leaner. The claim was that it can only mean an FR dip in broken gear is a false binary.
 
So does damping factor and its interaction with speaker impedance.
This has a direct influence on frequency response.
group delay.
This as well.
So does harmonic distortion spectrum, even at levels below typical audibility thresholds in isolated sine tests.
Let’s redo those tests with some music, see how well you do.
 
This has a direct influence on frequency response.

This as well.

Let’s redo those tests with some music, see how well you do.
You seem to be confusing frequency magnitude response with impulse response. These are functions of phase in the impulse response not frequency. GD is the negative derivative of phase.
 
You seem to be confusing frequency magnitude response with impulse response.

They are not independent. For causal systems, the impulse response completely determines both the magnitude and phase response. In minimum-phase systems, changing one necessarily affects the other.

These are functions of phase in the impulse response not frequency. GD is the negative derivative of phase.

Correct. But that doesn’t contradict what I said. In minimum-phase systems, phase, group delay, and magnitude response are mathematically linked. Electronics, loudspeakers, and rooms are not (always) perfectly minimum-phase, but neither are they arbitrary non-causal systems like linear-phase FIR filters where magnitude and phase can be adjusted independently.
 
Electronics, loudspeakers, and rooms are not (always) perfectly minimum-phase
Electronics, loudspeakers, and rooms are not (always) NEVER perfectly minimum-phase.
 
Electronics, loudspeakers, and rooms are not (always) NEVER perfectly minimum-phase.

Agree, but some clarification is needed. They key word is "perfectly". Minphase relies on an unspoken assumption - that it is linear time invariant (LTI). If it's not LTI, it can't be minphase, period. Electronics are more likely to be perfectly LTI, but not always. Push them out of the linear operating range and the LTI (and minphase) goes out the window. Loudspeakers, less so. Rooms, even less so such that it is dominated by non-minphase behaviour.
 
Electronics, loudspeakers, and rooms are not (always) NEVER perfectly minimum-phase.
They key word is "perfectly".

Real systems are neither perfectly LTI nor perfectly minimum phase, but many engineering approximations rely on the fact that they are sufficiently close over the frequency range and operating conditions of interest. That’s why the magnitude/phase relationship remains extremely useful even though it’s not exact. The point is that for causal systems there is generally a strong relationship between magnitude and phase, and for minimum-phase components it is mathematically fixed by the Hilbert transform. The existence of excess phase doesn’t invalidate that relationship; it just means it isn’t the whole story.

Otherwise, show me an amp that has a poor damping factor that does not influence the frequency response. Show me a practical speaker crossover that adds group delay without influencing frequency response...?
 
Live music, in person - always the gold standard though.
As an experience, definitely yes. The atmosphere, the audience reaction, the interplay between the artist and the audience (which includes the listener), the artist right there. Nothing beats it.

But the sound. Often way worse than a recording. The sound rattles round the venue. The PA stacks are way over there (Left) and there (right), the vocals are either lost of distorted. The drums are unbalanced.

And thats just rock and roll.

Much dance music has no 'live'. Its made in a studio, or a teenagers bedroom. The definitive experience is a dance venue, the DJ and sound man choose the Eq and volume that sound best to them, they add reverb, echo and even play about with the speed and pitch. But, for me the real version of reggae and Drum and Bass is a home made sound-system, the size of a van, spine re-adjusting bass and a sweaty dance floor. The bass would break my crockery and have the police round.
 
All pass filters.

Anyway I think we are in general agreement.
I knew this one would come ;) There is an important catch, though. Once you insert it into a practical crossover, its phase shift changes the summation between the drivers, so the acoustic frequency response changes as well. That’s exactly why all-pass networks are used for crossover alignment in the first place.

And as to the audibility of an all-pass filter on its own: you will have a very hard time unless the phase rotation is extreme.
 
And as to the audibility of an all-pass filter on its own: you will have a very hard time unless the phase rotation is extreme.
This is exactly why I consider controlled listening an essential complement to measurements. I spend a great deal of time linearizing crossover phase using allpass filters. The changes are hard to hear I agree, but one effect I repeatedly observe is that the center phantom image rises toward ear level as the drivers become better phase aligned. A vocalist that previously appears to originate near the floor often locks into a more natural height. Conventional measurements can show improved phase alignment, but they don't tell you how the auditory system reconstructs the resulting spatial image. Soundstage is ultimately a perceptual phenomenon created by the brain, so listening remains an indispensable part of evaluating these changes.

I appreciate that you may still disagree, and that's perfectly fine. I've said everything I wanted to say on the subject, so I'll leave it there.

Cheers!
 
This is exactly why I consider controlled listening an essential complement to measurements.
I think nobody will argue against that..
I spend a great deal of time linearizing crossover phase using allpass filters. The changes are hard to hear I agree, but one effect I repeatedly observe is that the center phantom image rises toward ear level as the drivers become better phase aligned.
I'm talking about ONLY an all-pass filter. As explained, once you combine this with the rest of the crossover, this will inevitably influence the frequency domain as well.
 
This is exactly why I consider controlled listening an essential complement to measurements. I spend a great deal of time linearizing crossover phase using allpass filters. The changes are hard to hear I agree, but one effect I repeatedly observe is that the center phantom image rises toward ear level as the drivers become better phase aligned. A vocalist that previously appears to originate near the floor often locks into a more natural height. Conventional measurements can show improved phase alignment, but they don't tell you how the auditory system reconstructs the resulting spatial image. Soundstage is ultimately a perceptual phenomenon created by the brain, so listening remains an indispensable part of evaluating these changes.

I appreciate that you may still disagree, and that's perfectly fine. I've said everything I wanted to say on the subject, so I'll leave it there.

Cheers!

I agree with you that listening is and should be an essential part when evaluating the things seen in the measurements, but I must say that I have never heard the effect you describe with the center phantom image appearing to come from a point lower than the height of the acoustic axis of the loudspeakers in use. Well, granted that you have made sure to level the speakers acoustic axis to the same height as the ear level at the main listening position.
 
All absolutely true. The important thing is that there is a difference between what you like and what you can measure. Personal preference varies for a variety of reasons. Measurement is the best decision support tool we have. It uses standardized devices that give reproducible results. Devices that produce very good measurements may not produce sound that all people like. Measurements do give us a way to look at the difference between input and output for everything in the signal chain. It's important to realize that the weakest links in the chain are the transducers, microphones and speakers or headphones. Electronics is pretty well solved, especially if you take human hearing thresholds into account.
There is no difference. If you can hear something, you can measure it. If you like a certain characteristic, you can measure that too. Precisely because the instruments we use (including ADCs) are better than human hearing by orders of magnitude.

You can then analyse what "character" you like and what it looks like in the measurements. Be thorough, eliminate your unreliable wetware between your ears as good as possible, and the conclusions will even enable you to buy blindly - or rather deafly - to a degree. It really works once you figured out what your taste technically, physically means.
 
Yes please lets see it!
I have flat (within +/-0.5dB) frequency response speakers and speech sounds incredibly realistic…. And I know (like all of us) what speech, unlike music is meant to sound like.

The importance of achieving the “flattest” frequency response as possible to get a more realistic representation of speech sounds is questionable.

The reality is that most microphones for recording is rather chosen for their characteristics more that “flatness”, and it’s not uncommon that the microphone is chosen by the vocalists based on how they perceive the sound of their voice ”in their head”. The loudspeakers in the studio used for mixing the speech weren’t necessarily very flat in the frequency response, and most of the time, there are some extreme equalization done to the vocal track to ”cut through” in the mix, and not being drowned out by the rest of the instrumentation.

All the above mentioned things makes the frequency ”deviation window” quite large for what will be perceived as a realistic and natural-sounding vocal track, and a loudspeaker with a flat +/-0.5dB frequency response will not necessarily sound more realistic than a loudspeaker with +/-3dB deviations in the response, which is quite common with most loudspeakers.
 
My main point is that things which measure identical sound identical.
I think I understand why some people get stuck on this. I agree in general with your OP, but I think this statement should be tempered a bit. The fact is, signals that *are* identical sound identical, and no one could argue otherwise. But merely to say they measured identical is not sufficient because it does not address whether the measurements performed were complete. The only way to be sure that two signals "measure identical" that I can think of would be to subtract them and show that the remainder is zero. This is a valid test and can easily be performed. But such strong conclusions should not be made based on the matching of a handful of commonly measured audio system properties alone, i.e., SNR, THD, SINAD, etc. These are necessary but not sufficient to claim identical. I think blind A/B testing is probably the best way to say if signals (or systems) sound identical. The fact that such testing tends to support the measurements above is really what gives them credence.
 
The main problem with the word 'sound' is that this is about sound pressure variations that are picked up by a microphone or ears.
An amplifier and DAC thus do not have 'a sound'. They can alter the applied signal in various ways though.
These are *easily* measurable and comparable. This 'easy' part more often than not is simply subtracting in and out signals.
A small phase or time difference can skew results. Nulling is not really easy.

Another thing is that even identical 'sounds' can still 'sound different' to someone simply because of the perception part which is variable.

Blind AB testing also depends on perception and part of perception is the level of training (detecting certain differences others may not pick up on).

But sure blind, level matched AB testing is the best we have but is not always easy.
Measurements are repeatable but as stated... the relevant measurements have to be made and must be done correctly.
Then there is another hurdle for both cases which is correct interpretation and statistics.
 
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@rjp Well, yes: Signals which are identical also measure identical. Which quantities like THD, SNR and others are required to safely determine that signals are identical is definitely open to debate. But a measurement is not just a number: Capturing the signal using an ADC is part of the measurement. THD is just a metric derived from that data. And if we captured a signal with high precision - often higher than the original recording we may have been playing - we are free to carry out any comparison we want with that data. Including direct comparisons using nulling or anything else.

My point is: People coming to ASR claiming that human hearing is superior to measurements often seem to think that there is some fundamental characteristic or some "magic" measurement instruments do miss when capturing a signal. That is simply not the case: These instruments are typically better than those used for the original recording.


@solderdude I would not be so strict as to say that amplifiers don't "sound" and only output electrical signals. Yes, that is technically correct. But I think that everybody understands the reality that in the end, we listen to amps through speakers or headphones. If they produce the same electrical signal which is fed to these transducers, they will produce the same sound under otherwise unchanged boundary conditions (speakers, room, environmental noise, and so on). Therefore, I'm OK to say that amps can and often do "sound identical".
 
@RandomEar What do you mean by “These instruments are typically better than those used for the original recording” and why is that relevant?
 
@RandomEar What do you mean by “These instruments are typically better than those used for the original recording” and why is that relevant?
ADCs designed and used for measurements are typically as good or better than ADCs used for recordings in the relevant measurable dimensions (SNR, THD, linearity, flatness of FR).

The idea is that they can definitely fully capture and analyze the characteristics of the device under test (e.g. an amplifier) and any potential audio sample / recording that device might play for the test. So nothing stays hidden from them - there can't be any magic property we miss when measuring, because our measurement instruments are at least as good as those used for the music recording itself.

That will, of course, not be true for all combinations of all ADCs ever used for measurements and recordings. But an Audio Precision analyzer or a Cosmos ADCiso is designed to be used to test and evaluate other ADCs or DACs - so it must outperform them in the relevant metrics, otherwise you couldn't use such an analyzer to judge or further improve the tested device.
 
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