• Welcome to ASR. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

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

RandomEar

Major Contributor
Joined
Feb 14, 2022
Messages
1,302
Likes
3,523
This post is intended as a reference for when the same question comes up the 1000ᵗʰ time and you don’t feel like typing out why the answer is “NO” again.

Short reminder: Decibel (dB) is a logarithmic scale. A difference of 20 dB in amplitude equals a factor of ten, which is also referred to as one order of magnitude.

Nope. Human hearing is vastly inferior to modern measurement instruments – in most dimensions by multiple orders of magnitude. It’s not even close. We can measure everything we can hear and much more than that on top.

This whole idea of ears somehow being superior is a non-starter, because the majority of the music we listen to and all the music audiophiles describe as “challenging” like classic music, jazz or female singers, has been captured using microphones and standard commercial analog-to-digital converters (ADCs) - which are measurement instruments. So the whole discussion could simply end here, because there is no counter-argument to this: If you think you can hear things which can’t be measured, then how the f*ck could those things ever be part of the recorded audio signal, which was generated by measuring it in the first place? As you can see, the rest of this post is technically redundant because the initial proposition has been debunked.

Let’s nonetheless look at how much better instruments are than ears, just to get a grasp of the amount of hubris which was required to come up with that idea. The following are some important quantities you could measure in an audio signal:
  • Level thresholds
    Humans hearing is limited to a range of about 0 dB SPL at 1 kHz (threshold of hearing) to 130±10 dB SPL (pain threshold) [1, 2]. Even consumer-grade instruments like a MiniDSP UMIK 2 reach about -30 dB SPL @ 1 kHZ in self-noise [3, 4]. Their upper limit is usually around 120-140 dB SPL, depending on the microphone type [5]. Technically, capturing even higher SPLs would be possible, but there are very few use cases where this would be required, which is probably why it’s uncommon for mics to support it.

  • Level differences
    Humans are actually pretty decent at hearing level differences as small as 0.25 dB for pure tones, with some people in some tests being sensitive down to about 0.1 dB [6, 7] (you can test it yourself here). However, there are multiple effects at play limiting that ideal case threshold during normal listening, like the hysteresis effect [8]. Measurement instruments are more precise than human hearing for level differences, with good analyzers reaching 0.03 dB [9]. But in practice, when measuring with microphones and outside of soundproof chambers, environmental factors like fluctuations in the room noise limit the achievable precision for humans and instruments alike.
    If you need to measure precise level differences on speakers or headphones, it is therefore essential to measure the driving voltage using a multimeter instead of measuring the SPL using a microphone or SPL meter. Even cheap multimeters measure down to 1 mV, giving you a much higher precision than any microphone (and also surpassing humans by a good margin).

  • Noise
    Since instruments can detect sounds at much lower levels than humans, the same is true for noise. As noise can usually be analyzed over longer time scales, whereas instantaneous values may be required for level differences, instruments can gain further sensitivity in detecting noise by increasing the analyzed time frame. Consequently, modern audio analyzers can detect noise at absurdly low levels (< -120 dBFS) [10, 11]. In an odd twist of fate, modern DACs have reached noise levels which are at or below those of the best currently available audio analyzers – but not better than the best lab-grade instruments like precision voltmeters.

  • Distortion
    Humans can hear distortion in real music down to about -40 dBFS (= 1%) with instant A/B switching [12]. For speakers, it is generally assumed that distortion below 0.3% or -50 dBFS won’t be audible [13]. However, with pure tones in very favorable (artificial) scenarios, some people can reach a threshold of close to -80 dBFS with very high order harmonics (>20th order) [14]. Even better consumer-grade ADCs like the E1DA Cosmos can easily detect distortion down at -140 dBFS, though, which is just insanely far away from anything any human could ever even dream of perceiving [15].
    The reason the threshold numbers for humans diverge so much is that THD and IMD are not well correlated to the audibility of distortion: Distortion components close to the original tone can be fully masked even at very high levels, while higher order distortion can be easier to detect due to the lack of masking [11, 16]. Nonetheless, absolute lower thresholds have been established for THD and IMD and are valid. Multiple psycho-acoustic models exist which correlate significantly better with the audibility of distortion [17]. But in practice, manufacturers and reviewers alike publish THD or IMD numbers: They are easy to measure with great precision and are the de facto industry standard.

  • Frequency thresholds
    Human hearing is usually accepted to be limited to a range of 20 Hz – 20 kHz for young listeners and the upper hearing threshold quickly degrades with age [18, 19]. But even if you assume that we might hear slightly above or below those limits, instruments can measure anything from 0 Hz to multiple 100 kHz for audio analyzers and into the MHz and even GHz range for oscilloscopes and spectrum analyzers [20, 21]. So, no chance for humans to compete.

  • Frequency differences
    Modern audio analyzers are precise to 10⁻⁵ Hz or better, whereas humans usually fail to detect differences below 1.0 Hz in test tones [10, 22]. There’s just no contest here.

  • Time precision (“Timing”)
    For better consumer ADCs capturing signals at 192 kHz and 24 bit, the intersample time is 5.2 µs. However, the actual time resolution is significantly higher due to the sampling theorem and clears the nanosecond range [23, 24]. In addition, different measurement instruments like consumer-grade oscilloscopes can easily display a fully resolved 50 MHz sine wave, which takes 20 ns for one full oscillation. This means the time resolution of the scope must be significantly better than that. For humans, the agreed upon number is somewhere between 7 and 18 µs, which is multiple orders of magnitude worse than instruments [25, 26].

  • Phase differences
    As phase differences are effectively time differences for specific frequencies, the same arguments concerning timing (and frequency) precision made above hold and human ears are far outclassed by measurement instruments.
This shows how much better instruments are in the most relevant dimensions of audio quality. But we are not quite finished.

Let’s also discuss some typical counterarguments that come up when talking about measurements:
  • But I can hear differences, even though the measurements say there are none!”
    There’s three options:
    • Your specific device(s) are defective and there actually are differences where there would be none in an undamaged device.
    • You did not account for all boundary conditions and tested incorrectly. For example you may have missed to level-match to within 0.2 dB (ideally 0.1 dB) or you are using audibly different filters, EQ or other settings. Your seating position may have changed, or you may have taken more than a couple of seconds to switch between the settings or devices you tried to compare.
    • You are mistaken and are a victim of bias, like any other human could be. Or in other words: Your brain is tricking you and the differences would instantly vanish in controlled (double) blind testing. This is by far the most common and prevalent error in audio testing when done by a layperson.

  • But there’s more to an audio signal than time and voltage/current!”
    No, there absolutely isn’t. You can watch everything which makes up the analog audio signal on a two-dimensional oscilloscope screen. There’s time, there’s amplitude, that’s it. As amplitude, you can either measure voltage or current. The current is a direct result of the voltage being present: If two audio signals have an identical voltage curve on a specific system, their current curve will also be identical.

  • Science hasn’t fully understood it, yet!” / “We discover new things all the time!”
    If that were true and even assuming that the science behind some basic electronics somehow were incorrect, you would still easily be able to prove that such a problem exists using well controlled double blind listening tests. As long as no such evidence exists, that whole assumption is flawed: You can’t postulate major errors in some of the most well understood fields of engineering without providing any evidence and expect people to take you seriously.
    The reality is that in relation to electric and acoustic signals (like analog and digital audio), science has understood these very well for over half a century. We – for the most part – also know what we don’t know. We have applied knowledge from all the fields of science and engineering related to these topic to build devices like acoustic metamaterials or phased array antennas. If there were any fundamental errors in the basic science behind it, these things would simply not work.
    The fact that an individual might not understand all the theories and calculations used to design a specific device does not invalidate the science behind it. On the contrary, if an opinion is uninformed, the opinion itself should be the first thing to be questioned.
With that being said, the fact that measurement instruments are superior to human hearing in every way doesn’t mean that we can perfectly interpret or link measured quantities to every subjective sensation a human might perceive. A couple of good examples for this usually come up with speakers: We can measure their frequency response and directivity and how they interact with a specific room. But from those measurements, it might still be difficult to tell if one specific speaker in one specific room might have a “forward” representation of voices or create a good representation of the “sound stage” encoded in your recording.

In a similar manner, old school tube amps often add copious amounts of distortion to the signal, which some listeners describe as pleasing up to a certain degree. We can precisely measure the distortion, but we might not be able to predict today what exact amount is perceived as pleasing and at what point it becomes annoying again. But we can say when it would be audible or inaudible. So for subjective qualities where we can pinpoint specific measured quantities as their source or threshold, we absolutely can predict how and if those will be perceived by humans.

The fact that we don’t know all links between subjective impressions and objective data yet also does not mean that two devices which measure identical in every way might still sound different under otherwise identical boundary conditions: They wont. There’s no “more identical” past “identical”. The only reasonable caveat here is the question if we measured all relevant quantities for a specific device to declare it “identical” to another one. The required measurements for this judgment depend on the type of device (DAC, amp, speaker, etc.) and the boundary conditions under which it is expected to perform. One number alone certainly won't cut it, but nobody around here argues that it will.


Note: If you happen to have better sources for some of the above bullet points, feel free to share them below. If I made a mistake, please let us know how and where.
 
Last edited:
ven consumer-grade instruments like a MiniDSP UMIK 2 reach below -30 dB in sensitivity
I think sensitivity is not the right word here. For the UMIK 2 the self noise is 19dB(A).
Only when doing long fft's you can 'see' lower than that.
 
Living in a decreasingly scientific world....I agree with the OP, and it's disappointing to see so many people just making it up and defending their right to do so. I could let people believe whatever they like but not when it comes to embracing BS like linking Autism to vaccines or declaring climate science a hoax, and Audio mythology is just a slightly less dangerous version of the same mentality.
 
Banger post. I'm partial to the "did you forget that recordings are also measurements" point myself.

This reminds me of the talk we got at the beginning of Econ class. They made a point of drawing a distinction between normative and positive statements, and tried hard to stay positive. In that sense it means factual observations, normative means opinionated on whether those facts are good or bad.

Measurements of audio equipment should be seen as positive. Whether they sound is good is normative.

Where discussions often go off the rails in the very beginning - people will come in with an opinion and assume there must be a fact to back it up, sometimes to the point that establishing a factual basis is seen as a distraction.

ASR goes in the other direction and waits for facts to be established before talking opinion. The person with the opinion is expected to come up with facts, too. Sometimes a rude awakening.
 
Even consumer-grade instruments like a MiniDSP UMIK 2 reach below -30 dB in sensitivity [3, 4].
This is highly misleading. The integrated self noise of the UMIK-2 is approximately 20dB(A), according to the datasheet. In a quiet room, a white noise source can be detected down to ~0dB SPL (Fielder, Louis D., Dynamic Range Issues in the Modern Digital Audio Environment). Good recording mics are typically quieter than the UMIK-2, but still very few have self noise below the threshold of hearing at all frequencies.
 
In full agreement with the OP.

Live music, in person - always the gold standard though.
 
Accompanying collection of poor counter argument’s:

 
This is highly misleading. The integrated self noise of the UMIK-2 is approximately 20dB(A), according to the datasheet. In a quiet room, a white noise source can be detected down to ~0dB SPL (Fielder, Louis D., Dynamic Range Issues in the Modern Digital Audio Environment). Good recording mics are typically quieter than the UMIK-2, but still very few have self noise below the threshold of hearing at all frequencies.
Maybe. In the context of this post, the question is "Can you detect a sound at that SPL with a mic". And you can definitely detect a sound in the noise floor down to a certain level. What would you say are the right metric and resulting lower limit here?
 
Surprisingly, most people are unaware of how much their own ears/brains change from hour to hour, and literally between the start of an album and the end.

In my limited testing on friends (or as some told me, "pranking") it is clear that all of their ears make rubbish measurement tools.
 
Nope. Human hearing is vastly inferior to modern measurement instruments – in most dimensions by multiple orders of magnitude. It’s not even close. We can measure everything we can hear and much more than that on top.
In this generality your statement is hardly defendable. One should know one's limits.
To the most part you are talking about the electrical and mathematical (digital) domain and there you are right.
In the world of dacs, amps. cables and the Audio Precision analyser we can measure much better than the hearing system is capable of after converting these signals to sound. It is no contest.

However, if you want to measure “what can be heard“, you have to compare in the acoustical domain. And there it is somewhat different.
It starts with something as simple as sensitivity.
Even consumer-grade instruments like a MiniDSP UMIK 2 reach below -30 dB in sensitivity [3, 4].
What is “-30 dB“ even supposed to mean? That is just a number (≈3%=0.03) in logarithmic “units“.
The ear is much more sensitive to low volume sounds than an UMIK (as others pointed out already) and that holds for most microphones. A mic that is more sensitive than the human ear at all frequencies is quite a fancy device.

For other characteristics as volume level or frequency response you are correct, microphones are much better to quantify differences than the ear+brain.
But to decide whether a FR-wiggle comes from interference or resonance already is not so easy, although it is "heard" differently.

And how about something a bit more complex?
How do you measure that there sometimes is a bit of an echo with some instruments from the right? Or that the woodwinds sound a bit further away with this speaker compared with that? How do “spaciousness“ or “image precision“ change with an additional diffuser/absorber in the back or toe-in of the speakers? What difference makes "stereo bass"?And so on.

If you want to measure ”what can be heard”, you would need measurement procedures that can catch that consistently.
Even if we can define and measure certain quantities (LOC, IACC) we still do not know too much about the impact of these on “what can be heard“. Acoustics is more complicated than an electrical signal and on top the question “what can be heard“ is a psychoacoustic one.
So the ear is at an advantage as you would need a model of the hearing system that is as good or better than the system itself.
We are not there yet.
 
I think I explained at length at the end of the OP that we can't yet link every subjective impression to an objective measure. So we agree.

I don't see a problem with that. My main point is that things which measure identical sound identical. In general, that limits it to HiFi electronics because no two speakers or headphones will ever measure truly identical. My main gripe is people coming here every second day explaining to ASR how stupid we are and that they can hear a difference in every DAC or amp or whatever they own. It get's old quickly.
 
This is highly misleading. The integrated self noise of the UMIK-2 is approximately 20dB(A), according to the datasheet. In a quiet room, a white noise source can be detected down to ~0dB SPL (Fielder, Louis D., Dynamic Range Issues in the Modern Digital Audio Environment). Good recording mics are typically quieter than the UMIK-2, but still very few have self noise below the threshold of hearing at all frequencies.
The self noise of Class 1 sound level meters (mic/pre-amp/slm system) that we use are stated at 16-17 dB. These are instruments from Norsonic and Bruel & Kjaer. The advice is that if you are measuring a sound level that is within 10 dB of the self noise then these factors should be taken into account in the uncertainty. It’s not a “you can’t do that” approach.
 
What is “-30 dB“ even supposed to mean? That is just a number (≈3%=0.03) in logarithmic “units“.
The sensitivity of a microphone is typically given as a range of values, for example the Norsonic 1225 field mic capsule has a sensitivity of -24.1 to -34.0 dB re V/Pa.
 
Without any alignment of the mentioned dB scales, the worst post ever!
 
Tangential note:

I often marvel at the audiophile vinyl fans and the incredibly elaborate, ridiculously expensive turntables that they claim open up ever new frontiers of "extracting more from the groove".

I have seen a Neumann cutting lathe in action, and it is a much, much simpler machine than any of these audio jewellery contraptions (though it is heavy and expensive), and the cutting head is moved tangentially across the lacquer.

Funnily enough almost all audiophile tables don't have a tangential arm and thus produce tracking distortion.
 
IMO dismissing consistent feedback from experienced reviewers and listeners on flagship products as mere bias is an oversimplification. Measurements are essential for understanding performance, but they don't ultimately determine whether a component is enjoyable to listen to. Many of the industry's leading manufacturers deliberately voice their products for the most satisfying listening experience, even when that means deviating slightly from what appears "perfect" on a measurement graph.

I appreciate ASR's emphasis on objective measurements, but the new Marantz Model 10 is an interesting example. It's an uncompromising, ultra high end, Class D amplifier developed by some of the industry's most accomplished engineers, backed by an enormous R&D budget and world-class component sourcing. Yet a number of independent reviewers have consistently described its midrange as leaner than the traditional Marantz Class AB sound.

That doesn't necessarily mean those impressions are correct, but it does suggest that today's standard measurements - frequency response, THD, SINAD, and similar metrics may not fully describe every aspect of perceived sound quality. Characteristics such as harmonic structure, dynamic intermodulation distortion, transient behavior, or complex loudspeaker load interactions may contribute to qualities like midrange body, tonal richness, or emotional engagement in ways that current measurement suites don't yet fully quantify.

This doesn't diminish the value of objective measurements - they remain indispensable. It simply acknowledges that, at least for now, carefully controlled listening tests are still the ultimate benchmark for assessing how enjoyable a product is in real-world use.
 
it does suggest that today's standard measurements - frequency response, THD, SINAD, and similar metrics may not fully describe every aspect of perceived sound quality. Characteristics such as harmonic structure, dynamic intermodulation distortion, transient behavior, or complex loudspeaker load interactions may contribute to qualities like midrange body, tonal richness, or emotional engagement in ways that current measurement suites don't yet fully quantify.
No
 
Back
Top Bottom