[reginalb]

Here is the answer to that question...
https://kenrockwell.com/apple/iphone-6-plus.htm#measurements
https://kenrockwell.com/apple/iphone-6s-plus-audio-quality.htm



One is a specific way of describing a problem and one is a non specific way that begs a subjective conclusion.

Who considers a 37.5Ω a low impedance load for a headphone amp? I'd call that a very average load. Not high, not low. Just average. I was actually surprised by the amount of roll for such a load.

 

[castleofargh]

I would ask what's being evaluated. Many claims here are whether one can hear a difference with any system. At first I thought it was DAC, then others chimed in with amp distortions, and also what kind of various perceptions our brains have at a given moment. I am a proponent of how our perceptions are always changing and can effect how we can "hear" something from day to day. The main "controlled" anecdote I have is the computer vs iPhone source through my Benchmark. My bias was that they'd sound the same: so I was taken aback that they sounded different.

clearly specifying what we're testing is very important! most of the time the test itself will need to adapt to what we're testing.
the idea that all "whatever" sound the same is not provable and as such can't be claimed. but it certainly is disprovable by demonstrating properly that 2 stuff are audibly different. even if only one such device exists and is only perceived as different by one guy on a full moon, that's enough to disprove the claim that they all sound the same. I don't think we should make global claims like those. better explain that failing to pass a blind test with DACs happens quite often. and that given the high level of fidelity achieved even by rather cheap DACs, it makes sense to expect mostly small to inaudible differences unless another issue somewhere is massive enough to manufacture it's own difference. like an aberrant level of noise from the source, or some electrical issue(ground loop or whatever), or maybe a severe gain mismatch between DAC and amp. those stuff do happen but they might not be considered as typical or proper use of the DAC. and if an unusual condition is necessary to get audible differences, it's debatable if we should conclude that the DACs sound different. for example many DACs have a different voltage output, that's a fairly audible difference right there. if that counts as audible difference between DACs, then, it's very easy to have 2 DACs that sound different ^_^. but I feel that as audiophiles we should be talking about more than changing the volume level. which brings me back to the necessity of proper testing conditions to avoid mistaking something for something else.


your anecdote about the input of the Benchmark dac doesn't look controlled to me in anyway. only that you expected the same and didn't get that impression, but that proves or disproves nothing and tells us nothing about the actual variation in sound or the true cause for it. to be clear I'm not saying you made it up, just that we're missing a form of control where you can compare your impressions to something(the control) to gain the ability to confirm if the difference was indeed heard. and also maybe that it was caused by the different sources, or by the adapter, or by using the different inputs on the DAC, or all of the above. right now even if you had the ability to confirm the audible difference in sound, I still wouldn't know what conclusion to draw from it. so it's the type of anecdote that could probably tell us something, but under uncontrolled conditions, what are we to do with it? make assumptions that because you expected something else then you couldn't possibly be victim of placebo? even that is arguable because your preconceptions about digital audio might not be strong enough to oppose your vision of using 2 different gears. vision isn't to be underestimated. after all, the visual cortex is the biggest one in our brain and it clearly tends to dominates over other senses when put to the test. so drawing conclusions from a sighted(here used for both meanings) experience is really not as easy as it might seem.

 

[castleofargh]

Nothing new is likely to be said in the revival of this thread, so I'll just repeat what I've said before: the issues of variability and unreliability of perception and memory which plague casual sighted listening comparisons are also a problem for blind comparisons.

There's no way of avoiding comparing a perceptual judgement of what's being heard with a memory of a perceptual judgment of what was heard before, since we can't compare two sounds at the same time. Blinding can remove expectation bias, but other problems of perception and memory remain.

It's a very different situation as compared to say doing a double-blind comparison of a placebo vs drug for blood pressure reduction, based on objective measurements of blood pressure which aren't affected by perception and memory issues. The "measurements" involved in auditory perception and memory are fundamentally subjective, not objective.

Summary:

- Casual sighted listening comparisons: can be quite unreliable, with potential for large differences perceived which aren't objectively there in the physical sound

- Blind listening comparisons with matching of volumes and music segments, and ability to switch rapidly: should be better than casual sighted listening comparisons, but could still be unreliable, especially for detection of small objective differences in the physical sound

without getting all philosophical, we are absolutely sure of nothing when it comes to objective reality. the unavoidable filter of our senses used to acquire any form of information, makes any view of objective reality an interpretation or at least an incomplete observation. looking for perfect answer in an area that will never provide it is just a waste of time and something used by lazy people to justify avoiding any effort toward trying to know a little more. science admits that nothing is perfect or final. even theories are open to change if new data is solid enough to suggest a need to revise it. and it's been done a bunch of times already.
so it's pretty obvious to me that a listening test isn't final. we can do many more and if at some point we learn how to pass or we simply had a good night of sleep in silence and our ears are well rested or whatever, that listening test will matter and change our conclusion of previously inaudible difference. I don't really see the issue about it. at one point it was decided that 0dB SPL was the threshold of audibility for a tone in the midrange, then we found that some people could perceive down to maybe -10dB, so we integrated that as new data and nobody went crazy about the impossibility of hearing below 0dB. but what is sure, nobody would have moved that threshold based on sighted impressions.

beyond the matter of everything being imperfect, rises the issue of basic requirements to demonstrate something. a blind test may provide such conditions. thanks to a control, some dependent and independent variables, the elimination of some undesired ones... all the notions and parameters helping to increase our confidence by comparing the resulting data to something specific to confirm the impact and the cause of impact. a sighted test usually won't provide any of this. personal impressions of a sighted test used to demonstrate a "fact" to someone else, can be summed up by "dude trust me". which is not going to be a very convincing demonstration unless I'm trying to convince my mom.

 

[Phronesis]

without getting all philosophical, we are absolutely sure of nothing when it comes to objective reality. the unavoidable filter of our senses used to acquire any form of information, makes any view of objective reality an interpretation or at least an incomplete observation. looking for perfect answer in an area that will never provide it is just a waste of time and something used by lazy people to justify avoiding any effort toward trying to know a little more. science admits that nothing is perfect or final. even theories are open to change if new data is solid enough to suggest a need to revise it. and it's been done a bunch of times already.
so it's pretty obvious to me that a listening test isn't final. we can do many more and if at some point we learn how to pass or we simply had a good night of sleep in silence and our ears are well rested or whatever, that listening test will matter and change our conclusion of previously inaudible difference. I don't really see the issue about it. at one point it was decided that 0dB SPL was the threshold of audibility for a tone in the midrange, then we found that some people could perceive down to maybe -10dB, so we integrated that as new data and nobody went crazy about the impossibility of hearing below 0dB. but what is sure, nobody would have moved that threshold based on sighted impressions.

beyond the matter of everything being imperfect, rises the issue of basic requirements to demonstrate something. a blind test may provide such conditions. thanks to a control, some dependent and independent variables, the elimination of some undesired ones... all the notions and parameters helping to increase our confidence by comparing the resulting data to something specific to confirm the impact and the cause of impact. a sighted test usually won't provide any of this. personal impressions of a sighted test used to demonstrate a "fact" to someone else, can be summed up by "dude trust me". which is not going to be a very convincing demonstration unless I'm trying to convince my mom.

I mostly agree, but still get hung up on the issue that we perceive at both conscious and subconscious levels, and someone may be able to register subconscious differences while being unable to consciously register them. Unless a test is set up to consider the subconscious aspect, the test could lead to false negative results. And to properly set up such a test, you need to have some understanding of how subconscious perception works. I know people would like it to all be simple and not deal with psychological aspects, but it's not simple and listening/hearing inherently involves psychological aspects.

 

[bigshot]

It's a good idea to minimize subconscious bias as much as possible. Blind testing, perhaps in comfortable surroundings with pleasant sounds is a good way to do that. Maybe you can't completely eliminate psychological reactions, but at least you can get close enough that you can figure out if a difference is audible or not.

 

[robthemac]

That guy who said he heard differences between DACs with an ABX box was great.

It's kinda like a physicist saying 'I have made some dark matter in my basement' then packing a sad when others ask for details.

 

[Phronesis]

It's a good idea to minimize subconscious bias as much as possible. Blind testing, perhaps in comfortable surroundings with pleasant sounds is a good way to do that. Maybe you can't completely eliminate psychological reactions, but at least you can get close enough that you can figure out if a difference is audible or not.

It's not just about bias, many factors can affect our auditory perception without our being aware of those effects, since most of the cognitive processing of auditory perception occurs at the subconscious level - we're just not consciously aware of that, because subconscious activity is largely hidden from us.

 

[KeithEmo]

I agree with something BigShot alluded to in a slightly later post...

A major part of the problem with discussing this sort of thing is that audiophiles often resort to inaccurate or vague terms to describe what they hear.
As a result, different people may use the same words to describe very different things, and it is often difficult to equate sound with measurements.
Also, much as some people prefer to disagree, exactly how certain measurable differences relate to AUDIBLE differences is still not always clearly defined.

To use the example I put forward as... an example...

To many people I know, Sabre DACs (especially those of the vintage of the 9018 chip), seem audibly to boost high frequencies.
SUBJECTIVELY, compared to DACs of many other brands, it sounds as if the Sabre DACs boost the upper midrange by about 1 dB.
(To offer the converse; if I take a component that uses an AD1955 DAC, and apply a 1 dB boost cenetered around 7 kHz, the result sounds more like an ESS9018 without the boost.)
However, when we measure the ESS9018 DAC, we find that its frequency response is exceptionally flat (like most other good DACs).
Therefore, an actual difference in frequency response DOES NOT account for the difference in sound... because no such measurable difference exists.
(Therefore, the difference in frequency response that people claim to hear does not actually exist.)
However, if we measure the impulse response of the filters used in the Sabre DACs, we find that they ARE in fact measurably quite different.
This would seem to suggest at least the possibility that the differences in filters, which we can measure, may account for the subjective/audible differences.

I'll also offer a more well known example of how sometimes it's simply a case of "not measuring the right thing".

Class B amplifiers commonly have a known flaw - known as "crossover notch distortion".
On an oscilloscope, this sort of distortion literally appears as a "notch" in the sine wave which appears right before and right after the zero crossing point of the waveform.
As it so happens, the actual distortion is largely independent of the amplitude of the music signal... for a given circuit, adjusted a given way, the notch will be a certain size.
As a result of this, the PERCENTAGE of distortion will vary depending on the signal level.
(With a large signal, the distortion will be a very small average percentage of the total; wheres a tiny signal may be almost entirely distortion.)
Basically, with a large signal, you have a very high level of distortion, but for a very small percentage of the time.
(In a poor design, with a certain input signal, you can quite literally have a signal with an AVERAGE THD of 0.5%, but with a THD of 20% for 0.5% of the TIME.)
Therefore, studies about the level of AVERAGE THD which is audible really do NOT apply to this situation (unless some controlled study has shown that they are similar).
(Many people seem to agree that excessive crossover notch distortion sounds "harsh"... but the amount which is required to be audible is not widely agreed upon.)

I'll offer another very simple example.....
Let's assume I have a speaker that exhibits ringing......
When I apply a 1 second burst at 1 kHz to this speaker, it continues to produce sound at -30 dB for 20 msec after the applied tone burst ends.
Now consider the THD of that speaker during the time interval from 5 msec to 15 msec AFTER THE APPLIED TONE BURST ENDS.
During that time interval, we have no input signal, but a clearly measurable output signal... so all of that output signal is "distortion".
So, for that time interval, the THD is 100%... but the AVERAGE THD is much lower (and it will be different depending on what arbitrary measurement interval you choose).

A similar situation exists with DACs....
The filters in DACs create some amount of ringing before and after transient signals.
Therefore, for some very short period of time, they produce an output signal that is "pure distortion".
The duration of this signal is very short, and most of the energy it contains is at "ultrasonic frequencies", but it also expends energy that is "borrowed" from the audible spectrum.
(When the signal is "spread out in time", much of the extra distortion is inaudible, but it also takes energy away from the "intended signal".)
Many people, including myself, believe that it is this ringing and its variants that accounts for the differences people claim to hear between different DAC filter choices.
And, as far as I know, very little actual controlled research has been done to confirm or deny this hypothesis.

Errrr.... yes and no.

Saying "plain old resampling of the signal" is an oversimplification.
In general, altering the sample rate involves some filtering.
If you look at the software you're using to convert between sample rates, you'll USUALLY find options for what sort of filtering to use (in R8Brain, or Izotope, you can choose both the sharpness and the cutoff frequency of the filter).
And, yes, you will hear slight differences in the output when you change those settings (those differences are usually pretty subtle).
And, yes, some of those differences involve audible differences in high-frequency response (which can affect "the tonal balance").
And, yes, some pros DO prefer one converter over another "because of how it sounds".

If you want to see a comparison of a whole bunch of sample rate converters.... check out this website:
http://src.infinitewave.ca/

You'll see that many sample rate conversion utilities do deliver virtually identical outputs...
But a surprising number actually DO cause significant alterations to the signal...

I would agree that, with a DAC, assuming that the data remains the same, then the resulting analog audio should sound identical.
However, apparently, on some DACs, the various TYPES of inputs have considerably different performance... particularly when it comes to rejecting jitter present on the incoming source.
(Feel free to argue for another ten pages about whether the difference should be audible or not... but you can readily measure it.)

I would also add that, subjectively, many people I know agree that Sabre DACs tend to emphasize high frequencies, resulting in a sound that seems more detailed, yet slightly unnatural. I'm not at all interested in arguing about "how we must be imagining it" - nor do I know anyone who has done proper tests to reasonably confirm or deny it. HOWEVER, in their early product literature, ESS (who makes Sabre DACs) claimed that, rather than design their DACs for the most accurate response, they used focus groups to "choose the output filter that most people preferred the sound of". They at least claimed, at that point, not that their DACs were accurate, but that they "sounded different and better". They have notably omitted this claim in recent years.

 

[KeithEmo]

My expectations would also be that they would sound pretty much the same.

Various sources, like an iPhone and a computer, may have slight differences... like more or less digital jitter on their output signal.
However, like many modern DACs, the Benchmark DACs have internal mechanisms designed to eliminate most of these issues.
(Regardless of which source may start out with more jitter, or whether jitter is audible or not, the Benchmark DAC is designed to be essentially immune to it.)

I would be inclined to suspect that any significant differences were actually due to differences in the signal itself not being identical.
It's possible that, depending on the player software you were using, one or both were not delivering an unaltered (bit perfect) data stream.
This is true for many player programs.
However, it is even more often the case with streaming audio.
For example, when playing MQA files, the Tidal client for a computer does very different processing than the Tidal client for a phone or device like a Sonos box.
Also note that adjusting the volume on either the phone or the computer requires a modification of the bit stream...
The algorithms used to do so may be different on a phone than on a computer - and either may introduce other differences while doing so.
(So, when doing that sort of comparison, you must set both digital devices at "100% volume" and use the volume control on the DAC.
And, even then, you have no assurance that the software is really providing a pure unaltered signal at 100% volume setting. )

My original goal was to see if using the iPhone for my source would be a good system, or if I needed to make it larger with having a computer. My conscious expectation was that the sources would sound the same (as they both were using the same USB adapter going to the Benchmark's coaxial port). My focus at that point of time was just listening to the same track of music (there wasn't a preconceived notion or change in stimuli).

@Zapp_Fan: there definitely was a difference in tonality between the computer and iPhone going to my DAC. On further reflection as to why that is: perhaps the source OS is applying a DSP with the USB adapter (as its seen as a soundcard). Maybe it's just a matter of things being more complicated and the DAC not getting an unaltered bit for bit copy of the file.

 

[analogsurviver]

I agree with most of what @KeithEmo said - the most audible differences always lead back to transient response in any sound (re)production device. The mechanisms in either mechanical or electrical part of the equation may well be different - but they manifest themselves rather similarly in audible sound. It is a part of my striving - to quote @castleofargh - for *moARR HIGHS* (end quote) - as larger bandwidth always goes hand in hand with lower problems of "stored energy" ( be it in literal sense when electromechanical or vice versa transducers are involved ) , filter defficiences, etc, etc. Steady state frequency response can be stellarly flat from 20-20k - yet the device in question may - or may not - (mis)behave ( often quite BADLY ) outside this range - both below and above "band of interest". For me, a device having a response +- 2 dB 20-20K but not misbehaving outside this range is FAR better than something +-0.01dB 20-20K and going berserk just above that - be it resonance in an analog phono cartridge or brickwall filtering of a digital device - most commonly that would be RBCD .

I do not have either time or will to respond to every word and twisting of that same word with those who claim there are no audible differences. I can even sympathize with them - it may actually be so with their equipment. If they did any real work by themselves, they would have known that a SINGLE "wrong" electronic part ( capacitor, resistor, semiconductor, etc ) can be a make or break of the entire system - or audio chain, if you prefer it that way. Having literally hundreds of those in "transparent" audio equipment will surely render any really good device or recording a moot point - the difference simply can not get trough all those *filters/compressors* that are built in most of the audio equipment, studio gear here being the bigger culprit than better home audio equipment.

That insistance of furnishing the proof ... OK, I agree it is correct. Now... WHO has the finance to support a really meaningful test ? Like furnishing say 40 or more pairs of TOTL headphone sets, say at least 2-3k a pop ? I regret I did not bookmark the reply to the 1M challenge to that person who would be able to scientifically prove that cables do introduce audible differences - the COST of doing so, in order to satisfy all the criteria, would exceed that 1M ... - before you know it. The proponents of maintaining status quo have this gamme rigged so THOROUGHLY that whatever one does to change it, he/she will lose ... proving say 1M worth of claims "beyond the shadow of a doubt" could run one ten times or more in excess of that 1M.

 

[Arpiben]

I agree with something BigShot alluded to in a slightly later post...

A similar situation exists with DACs....
The filters in DACs create some amount of ringing before and after transient signals.
Therefore, for some very short period of time, they produce an output signal that is "pure distortion".
The duration of this signal is very short, and most of the energy it contains is at "ultrasonic frequencies", but it also expends energy that is "borrowed" from the audible spectrum.
(When the signal is "spread out in time", much of the extra distortion is inaudible, but it also takes energy away from the "intended signal".)
Many people, including myself, believe that it is this ringing and its variants that accounts for the differences people claim to hear between different DAC filter choices.
And, as far as I know, very little actual controlled research has been done to confirm or deny this hypothesis.

If your input signal is perfectly bandwidth limited in theory you should not have any aliasing whatever digital reconstruction filter applied inside DAC.
But nothing is perfect,I know...starting from input to windowing effects at boundaries and many others...
In principle,nowadays it is quite simple to measure the analog output of a DAC with large bandwidth digital instruments and proper input signals.
It is also quite easy to measure the differences of DAC's output when upscaling the input with different filter types ( million or not million taps).
I am not sure to have fully understood your "borrowed energy" explanation, anyhow the loss can be compensated by applying correction filers if necessary.
I am not trusting my ears but I do trust a measurement if properly done and conducted.

 

[JRG1990]

Therefore, an actual difference in frequency response DOES NOT account for the difference in sound... because no such measurable difference exists.
(Therefore, the difference in frequency response that people claim to hear does not actually exist.)
However, if we measure the impulse response of the filters used in the Sabre DACs, we find that they ARE in fact measurably quite different.
This would seem to suggest at least the possibility that the differences in filters, which we can measure, may account for the subjective/audible differences.

I'll also offer a more well known example of how sometimes it's simply a case of "not measuring the right thing".

Class B amplifiers commonly have a known flaw - known as "crossover notch distortion".
On an oscilloscope, this sort of distortion literally appears as a "notch" in the sine wave which appears right before and right after the zero crossing point of the waveform.
As it so happens, the actual distortion is largely independent of the amplitude of the music signal... for a given circuit, adjusted a given way, the notch will be a certain size.
As a result of this, the PERCENTAGE of distortion will vary depending on the signal level.
(With a large signal, the distortion will be a very small average percentage of the total; wheres a tiny signal may be almost entirely distortion.)
Basically, with a large signal, you have a very high level of distortion, but for a very small percentage of the time.
(In a poor design, with a certain input signal, you can quite literally have a signal with an AVERAGE THD of 0.5%, but with a THD of 20% for 0.5% of the TIME.)
Therefore, studies about the level of AVERAGE THD which is audible really do NOT apply to this situation (unless some controlled study has shown that they are similar).
(Many people seem to agree that excessive crossover notch distortion sounds "harsh"... but the amount which is required to be audible is not widely agreed upon.)

I'll offer another very simple example.....
Let's assume I have a speaker that exhibits ringing......
When I apply a 1 second burst at 1 kHz to this speaker, it continues to produce sound at -30 dB for 20 need

A similar situation exists with DACs....
The filters in DACs create some amount of ringing before and after transient signals.
Therefore, for some very short period of time, they produce an output signal that is "pure distortion".
The duration of this signal is very short, and most of the energy it contains is at "ultrasonic frequencies", but it also expends energy that is "borrowed" from the audible spectrum.
(When the signal is "spread out in time", much of the extra distortion is inaudible, but it also takes energy away from the "intended signal".)
Many people, including myself, believe that it is this ringing and its variants that accounts for the differences people claim to hear between different DAC filter choices.
And, as far as I know, very little actual controlled research has been done to confirm or deny this hypothesis.

What about upsampling dacs that have different filters and the filtering is done well outside the audiable range?.

Also transducers will likely have alot more ringing than a dac, so wouldn't it mask any ringing from the dac?.

 

[KeithEmo]

Quite so.....

The "borrowed energy" idea is more way of looking at the situation.

Assuming I have an input signal and an output signal.
And assuming that, when I compare them, their overall average energy measures to be absolutely identical.
(This is what "a totally flat overall frequency response" would demonstrate.)
But, when I look at the output signal with an oscilloscope, I have ringing that wasn't present in the input signal.
The energy that is present in the ringing must have come from somewhere.
If it was added from some outside source then the overall total amount of energy would be higher.
Since the overall amount of energy is NOT higher, the energy in the ringing must have been "borrowed" from some other point in time.
Therefore, even though the time-averaged frequency response is flat.
We would expect to see ripples in the frequency response at various points in time.

Likewise... there is no such thing as a perfect filter.
Assuming that your input content actually CONTAINS nothing above the Nyquist frequency, then all required conditions would be met without filtering, and it would all work out "perfectly".
However, since there is ALWAYS background noise, some of which is above the Nyquist frequency, there is always something that must be filtered out.
However, no filter is or can be "perfectly sharp"; an infinitely sharp filter would have infinitely poor time response; and a filter with infinitely perfect time response cannot be a filter.
In fact, the electronic components from which the filter is built will also contribute some noise and distortion.
Therefore, in practical terms, there can never be "a perfectly band limited signal", and there can never be "a perfect reconstruction filter".
You can get REALLY close... but it can never be perfect.
So we end up right back at "the limits of audibility"... and deciding what's "so far below those limits that it really need not be considered".

The bottom line is that the differences between the outputs of DACs using various filters CAN easily be measured.
For example, take the same model of DAC, and choose two different output filter options.
(Most manufacturers of DAC chips provide graphs which show the transient response of their various filter options.)
In some cases, you will find that the options sound audibly different; and, in most cases, you will find that the output signals are visibly different on an oscilloscope, and will NOT null to zero.
(Most people can only identify differences of several percent visually on an oscilloscope display; but smaller differences can be resolved by other measurements.)
So, we now have measurable differences, and claims of audible differences....
Therefore, the challenge is to correlate them.

If your input signal is perfectly bandwidth limited in theory you should not have any aliasing whatever digital reconstruction filter applied inside DAC.
But nothing is perfect,I know...starting from input to windowing effects at boundaries and many others...
In principle,nowadays it is quite simple to measure the analog output of a DAC with large bandwidth digital instruments and proper input signals.
It is also quite easy to measure the differences of DAC's output when upscaling the input with different filter types ( million or not million taps).
I am not sure to have fully understood your "borrowed energy" explanation, anyhow the loss can be compensated by applying correction filers if necessary.
I am not trusting my ears but I do trust a measurement if properly done and conducted.

 

[KeithEmo]

The problem is that we're approaching dangerously close to circular logic here....
You cannot argue that something is "well outside the audible range... because you must not really be hearing it... because it is outside the audible range".

There are audible differences between some filter choices offered on some DACs - even when the overall frequency response measures close enough that we would consider the differences to be negligible.
Either we're wrong about what we consider "inaudible"...
Or we're wrong about what we consider "identical withing negligible amounts of differences"...
Or there's something else going on that we've so far missed entirely...

I would just point out that "being sure there can't possibly be an audible difference" is itself a type of very powerful expectation bias...
Therefore, while I would agree that I would not accept the results of subjective experiences as facts...
Neither will I accept many arguments that claim to know with absolute certainty that those results are impossible...

Likewise, I would EXPECT the amount of ringing present in a tweeter to thoroughly mask the ringing in a DAC.
However, I don't know for sure how well such masking might work.
Perhaps it explains why I hear certain things with electrostatic headphones (which have very little ringing) and not with speakers.
I personally find quantum mechanics to be "intuitively ridiculous" - but, apparently, in reality it is not.

What about upsampling dacs that have different filters and the filtering is done well outside the audiable range?.

Also transducers will likely have alot more ringing than a dac, so wouldn't it mask any ringing from the dac?.

 

[Zapp_Fan]

Apparently filters can ring for dozens of milliseconds, so it's not a sure thing that your transducer is ringing longer than your filter at any given frequency. A really good pair of headphones should have decay times in a comparable range. Filter ringing will probably be a lot lower in amplitude, but I wouldn't say it's impossible to hear or that there are cases where filter ringing can't matter.