I am reminded that our biology would already reflects the trends, since it would be directly manipulated by them and that the loudness curve would be the result....
This would be trending the that has carved into our biology.
This would be the resulting sensitivity as our Biology seeks balance and survival.
When it comes to perception, I don't think we are lacking in bass spectrum. We are simply focusing on the ear when in fact, perception is being handed off to other areas of input.
Grok says
"
Where ears falter in LF, our biology leans on mechanoreceptors in skin, muscles, and bones—here’s how:
Answer: Yes, where ears lose LF sensitivity (<100 Hz), tactile senses—skin, bones—pick up 20-100 Hz vibrations, filling the gap evolution left."
This would be trending the that has carved into our biology.
This would be the resulting sensitivity as our Biology seeks balance and survival.
When it comes to perception, I don't think we are lacking in bass spectrum. We are simply focusing on the ear when in fact, perception is being handed off to other areas of input.
Grok says
"
Where ears falter in LF, our biology leans on mechanoreceptors in skin, muscles, and bones—here’s how:
- Frequency Range Overlap:
- Ears: 20 Hz-20 kHz, but <100 Hz needs high intensity (e.g., 40 dB SPL at 50 Hz, 60 dB at 20 Hz) to perceive.
- Tactile: Skin senses ~10-250 Hz via Pacinian corpuscles (deep vibration) and Meissner’s corpuscles (surface flutter)—peaks at 20-100 Hz, exactly where hearing weakens.
- Evidence: Studies (e.g., Verrillo, 1963) show tactile thresholds drop to ~0-10 dB SPL equivalent at 20-50 Hz—skin “hears” LF better than ears at low levels.
- Evolutionary Pressures:
- Predation (~400 Mya - 300,000 Ya): LF rumbles (e.g., footsteps, thunder) were felt through ground or air—tactile cues warned when ears couldn’t resolve direction (binaural cues fail <200 Hz).
- Environment: Vibrations from wind or prey movement—body sensing endured as a backup to LF’s auditory neglect.
- Result: Tactile system evolved sensitivity to <100 Hz, compensating for cochlear limits (coarse 10 Hz resolution at 20-100 Hz).
- Mechanisms:
- Skin: Pacinian corpuscles detect 20-100 Hz vibrations—e.g., 30 Hz at 10 µm displacement (~20 dB SPL equivalent) feels like a pulse.
- Bones/Joints: Low-frequency sound conducts through skeleton—20 Hz at 60 dB SPL vibrates chest or skull, bypassing ear’s threshold.
- Nervous Integration: Brain merges tactile and auditory input (multisensory neurons in auditory cortex)—LF “feel” enhances perception (e.g., bass in concerts).
- Practical Impact:
- Thresholds: Hearing needs 40 dB at 50 Hz to detect; tactile senses it at ~10-20 dB via skin—gap closed.
- Localization: Ears struggle with LF direction (<200 Hz, interaural time maxes out); body feels source proximity or intensity instead.
- Experience: Subwoofers at 20-30 Hz—ears hear faintly, but chest thumps—tactile dominates.
Why Tactile Steps In
- Physics: LF wavelengths (e.g., 17 m at 20 Hz) dwarf ear structures (pinna ~5 cm)—tactile uses whole body as a receptor.
- Survival: Feeling LF warned of distant threats (e.g., stampedes) when hearing lacked detail—biology doubled down.
- Efficiency: Enhancing cochlear LF sensitivity (e.g., bigger basilar membrane) was costlier than leveraging existing mechanoreceptors.
Result
Our auditory H(f) H(f) H(f) sacrifices LF sensitivity (high thresholds <100 Hz)—tactile sensation compensates, extending perception to 10-100 Hz via skin and bone, where ears fail.Answer: Yes, where ears lose LF sensitivity (<100 Hz), tactile senses—skin, bones—pick up 20-100 Hz vibrations, filling the gap evolution left."
It takes time to acclimate to quality and get free of our biases for what we have heard before.
I think there are several levels acclimation. One is the happens on mechanical side of things where your ears mechanism are already acclimated to a signature and then when change happens, it stands out. Like eyes acclimated to the dark and then brought into broad daylight. The other are psychological where there is a period before our brain says, eventually, "the sound is going to sound like this" and then eventually "the sound is supposed to sound like this" basically levels of expectations that are eventually accepted into the psyche. Its more complicated than that, figuring in all the other influences versus how far away from balanced tone we are talking about auditioning but In my experience it takes at least the better part of a weeks worth of daily listening sessions to really experience accept a voicing.To a person who is used to listening to bass boosted FR, of course a near flat response sounds off... Its like a person coming from a dark room, no light, eyes adjusted... then walking outside into a full Sunlit day... Their eyes will be in pain... but is it too bright outside? No... Our ears work similar, it takes time to adjust... and even then it takes time for expectation of the sound coming out of the loudspeaker system to change.
It's a bit of a cliché, but when you listen to speakers and start analyzing all sorts of things, or your attention is drawn to specific aspects of the (reproduction of) music, there is usually something wrong.
Quality monitor speakers are also useful for casual listening.
Quality monitor speakers are also useful for casual listening.
You said..Yes, I'm not talking about guitar cabs.
It's apparent you were talking about the speaker having these qualities, since it goes without saying that they are already present in the recording.However, these characteristics should be present, but subtle and ideally in a homogeneous whole.
A speaker does not need these qualities in order to successfully reproduce a recording which does.
P.S. There is some anecdotal evidence which confuses this issue (the subjective term 'analytical' comes to mind). These stories which involve acoustical problems shouldn't be confused with what's being said here.
I think this largely depends on the frame of reference.I think there are several levels acclimation. One is the happens on mechanical side of things where your ears mechanism are already acclimated to a signature and then when change happens, it stands out. Like eyes acclimated to the dark and then brought into broad daylight. The other are psychological where there is a period before our brain says, eventually, "the sound is going to sound like this" and then eventually "the sound is supposed to sound like this" basically levels of expectations that are eventually accepted into the psyche. Its more complicated than that, figuring in all the other influences versus how far away from balanced tone we are talking about auditioning but In my experience it takes at least the better part of a weeks worth of daily listening sessions to really experience accept a voicing.
For me (and my brother) it's usually immediately clear whether reproduction is good, mediocre or (seriously) deficient.
This doesn't imply 'perfection', because that doesn't exist. It's about 'credibility' for lack of a better term.
It kind of depends on the music of course.
Scale also matters.
A Kii Seven is a decent monitor speaker, but evidently limited by its size.
And although ATC makes low-sensitivity (monitor) speakers, I understand why they are popular among producers/mastering engineers.
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This is true only when listening to accurate monitoring, though I guess inaccurate monitoring is the same as a bad tone for sound engineer except we have to assume that the content is of good tonal quality.... and it last for only as long as you can keep your ears fresh.... I have a article someone where on my computer that relates the mechanism of the ear to wide band compression and dynamic Eq, where the longer you listen the more and more you press into the threshold effectively compressing the input. The mechanism of the ear seek neutralize tone as according to its transfer function and it seeks to lower levels on a wide band scale, so for example, listening to loud music over time, it will not seem as loud, as it did when you ears were "fresh". In the same manner you will lose sensitivity or resolution to tone as the ear fatigues, with level and duration. This is something that is very important for a sound engineer and I have come up with ways to keep the ears fresh. Lower volumes and short durations of monitoring follower by periods of silence then repeat. There are many nights the engineer EQs his/her heart out after production and mixing duties carry them on for hours, only to hear the blemishes gone untouched when they return to the project in the morning with fresh ears.It's a bit of a cliché, but when you listen to speakers and start analyzing all sorts of things, or your attention is drawn to specific aspects of the (reproduction of) music, there is usually something wrong.
Well I am not certain what deficit exist among low sensitivity speakers, as it is possible for them to still avoid signal compression at domestic levels and mechanical efficiency is still fine with large ATC having a 15" woofer and a 3" dome in a 6" waveguide for its career carrying midrange that has great decay performance.although ATC makes low-sensitivity (monitor) speakers, I understand why they are popular among producers/mastering engineers.
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Although I have several class D amps myself and I admire the contributions of Bruno Putzeys, my conclusion was that Class D doesn't meet the most stringent requirements for signal integrity.
Would that be from the point of view of the PCB design -> to ensure the signals, a class D amp has to process, preserve the overall integrity... or..
.. would that be related to the (complex design) nature of the class D amplification that manipulates the analog signal, irrespective of how good the PCB design is?
Or a combination of the two above..?
It's possible many don't know what they'd really prefer because they haven't heard it yet.
I think it’s less about knowing or not knowing and simply a matter of liking or disliking, or being familiar or unfamiliar.
Yeas ago, I once visited a mansion of an audiophile (and a musician) who was much older than me and was surprised at how far the frequency response of his old Tannoy speakers deviated from the standard, but it didn’t seem to be an issue for him at all. Given that he had also keep checking modern speakers, he must have been aware of the difference. I still have great respect for the speaker he chose as the best over the course of his long life, and there was a certain persuasiveness to the sound. When discussing audio as a hobby, I believe whether something is scientifically correct or not is mostly irrelevant, and looking down on others for appreciating something scientifically "incorrect," as many ASR followers do, is actually a greater display of ignorance.
They just don't know.
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I know that Tannoy has a reputation and there's potential in the typical Tannoy configuration.
Within reason, whatever you do to the response won't change the fundamental character of the speaker. I'm not just talking about the consistency and flaws of the speaker but how it plays out into the room.
That may help to explain some of the EQ efforts people employ. It's also meant to put on-axis response into perspective. I hear it said that on-axis response is king.. but it's a guideline, not a mantra.. unless we want to ignore what's been learned in recent decades. If you get the rest of the speaker right and fitting for the room, the listening axis tends to fall into line...
Within reason, whatever you do to the response won't change the fundamental character of the speaker. I'm not just talking about the consistency and flaws of the speaker but how it plays out into the room.
That may help to explain some of the EQ efforts people employ. It's also meant to put on-axis response into perspective. I hear it said that on-axis response is king.. but it's a guideline, not a mantra.. unless we want to ignore what's been learned in recent decades. If you get the rest of the speaker right and fitting for the room, the listening axis tends to fall into line...
I have only ever once "voiced" a loudspeaker based on listening tests. That's when I concluded that a flat axial response CD waveguide will sound too bright and so I built in a subtle taper at the high end. But we know why this needs to be done and that is because the HF power response of a CD waveguide is far greater than that of a piston source thus yielding a brighter sound in comparison.Voicing should first start with close measurements, is what I think I know. Nearfield measurements. I think. Here, I think starting with a neutral response makes sense. What ever filters needed to get the desired response is ok at this stage. After we have done good work here, then one would move to the listening position. From past discussion, I think it was said that in the areas of the spectrum where there is high amounts of comb filtering, eqing should be limiting to low Q filters reflecting 1/3 octave or less. I don't know if I am saying that right. Basically, changes should be as resolved as what you see with 1/3rd smoothing but this approach is reserved to where comb filtering is obvious. Beyond that, I think possibly, anything goes? I know that in LF its ok to EQ with high resolution, possibly sharp filters should be avoided at some rate, increasingly as we move up the spectrum, possibly governed by comb filtering.
How far off am I @gedlee
Beyond that all I ever do is to make the listening axis as flat as possible while maintaining as smooth, flat and high a DI as possible. This is done using only data above about 200 Hz which is easily obtained in a large room at 1 meter distance. I just use my living room. I never even look at the response below 200 Hz until the system is in-situ. Then subs are added and adjusted for as smooth a gradual rise below 200Hz as I can obtain (in a reasonable amount of time.) Nothing is ever changed based on listening. I stand by and swear by this technique. Everything else sounds deficient to me.
Yes, tactile sensation at LFs is very important. That's why I pay attention to the furniture that I am sitting on in my listening room because resonant furniture can create an undesirable LF sensation - think butt shakers.Answer: Yes, where ears lose LF sensitivity (<100 Hz), tactile senses—skin, bones—pick up 20-100 Hz vibrations, filling the gap evolution left."
Sure the "bad" can often be obvious. It's when trying to evaluate a really good system that immediate judgments are not likely to be accurate.For me (and my brother) it's usually immediately clear whether reproduction is good, mediocre or (seriously) deficient.
I have many musician friends many professionals with awards and multiple CDs. They are not very good sound system evaluators. They only hear the musical performance and completely look past and reproduction flaws.Yeas ago, I once visited a mansion of an audiophile (and a musician) who was much older than me and was surprised at how far the frequency response of his old Tannoy speakers deviated from the standard, but it didn’t seem to be an issue for him at all.
In one case I pointed out a passage on one of his CDs that was very clearly clipping. He couldn't hear it although everyone else could. He could only hear how he played the passage not how it was reproduced.
I almost never trust musicians judgements of sound quality of reproduction.
Indeed, most musicians have no clue, also because they hear live music different when they play than we, the public.
I advised some quiet known classical musicians in what speakers to buy, and let them listen to a few very different sets and most did not hear the difference. As long as it's acceptable sounding and not fatiguing they don't care in most cases. Those classic musicians bought Neumann KG420G's (for one space) and Meyer Bluehorn system (for a bigger space) and both are very happy with that. Those are very neutral sounding systems and are way better (in their opinion) than the very coloured Kiplish speakers they both had before. They hire me because they know me and my knowledge from before and that i'm not connected to a shop (i got paid the same, whatever they bought). In a shop they would have had whatever the fashion of the moment was because they have no clue.
Off course you also got musicians who are also audio freaks with knowledge, but it's like everywhere a minority.
I advised some quiet known classical musicians in what speakers to buy, and let them listen to a few very different sets and most did not hear the difference. As long as it's acceptable sounding and not fatiguing they don't care in most cases. Those classic musicians bought Neumann KG420G's (for one space) and Meyer Bluehorn system (for a bigger space) and both are very happy with that. Those are very neutral sounding systems and are way better (in their opinion) than the very coloured Kiplish speakers they both had before. They hire me because they know me and my knowledge from before and that i'm not connected to a shop (i got paid the same, whatever they bought). In a shop they would have had whatever the fashion of the moment was because they have no clue.
Off course you also got musicians who are also audio freaks with knowledge, but it's like everywhere a minority.
Both, but mainly due to the modulation of the original signal and the additional circuitry required (I've attached a paper including an overview).Would that be from the point of view of the PCB design -> to ensure the signals, a class D amp has to process, preserve the overall integrity... or..
.. would that be related to the (complex design) nature of the class D amplification that manipulates the analog signal, irrespective of how good the PCB design is?
Or a combination of the two above..?
A simple example:
It's not that I'm biased towards 'new' technology because of nostalgia.
My brother and I still have a few of the first (pre-hypex) class D amplifiers, which also contain early R&D input from Bruno Putzeys.
The modulator is one of the last audio ICs from Philips NatLab.
Those amps are now 20 years old.
I also own modern class D amps.
For most people class D is sufficient, as are ΔΣ DACs.
I'm not trying to convert anyone, I'm just sharing my own opinions, based on experience and study.
My focus goes beyond (standard) measurements alone.
Knowledge, opinions, preferences and viewpoints differ.
An example regarding loudspeakers, in light of my replies to Dr. Geddes.
Dr. Geddes uses B&C 15NBX100-8 for his NS15 speakers.
Factory impedance of this woofer:

Looks pretty clean/well-behaved below 1000 Hz.
(Usable) SPL of the same woofer:

It's a pure woofer, not a midwoofer. It's also quite popular for PA subwoofer applications due to the generous Xmax and power handling.
This is the impedance of another B&C woofer (that I would prefer):

Some ripples throughout the whole midrange.
Usable SPL:

Takeaways:
The 15NBX100 is less likely to reproduce midrange frequencies accurately compared to the second woofer, due to trade-offs in its design for high-excursion bass performance. Based on T&S parameters:
- Moving Mass (Mms = 151 g)
- The higher moving mass (151g vs. 93g) reduces transient response speed, making it slower to start/stop during rapid midrange signals like vocals or guitar plucks
- This mass penalty is amplified by the lower compliance (CMS = 116 µm/N vs. 170 µm/N), requiring more force to move the cone.
- Voice Coil Inductance (LE = 2 mH)
- Higher inductance causes impedance to rise sharply at higher frequencies, attenuating midrange output above ~500 Hz
- The second woofer’s lower inductance (0.5 mH) minimizes this issue, preserving midrange clarity.
- Motor Strength Trade-offs
- While the stronger motor (Bl = 25 N/A vs. 19.8 N/A) improves cone control, the benefits are offset by the heavier cone.
- The motor constant (Bl/√Re = 11.1 N/√W) prioritizes efficiency over midrange detail compared to the second woofer (8.68 N/√W)
- Resonance Frequency (Fs = 36 Hz)
- The slightly lower Fs shifts the driver’s natural resonance deeper into sub-bass territory, reducing midrange energy in the 80–300 Hz range where many instruments reside.
The 15NBX100 is engineered for high-excursion sub-bass performance (10mm Xmax, 1000W RMS) at the expense of midrange clarity. Its heavy cone and high inductance make it better suited for:
- Dedicated subwoofer applications
- High-SPL bass reinforcement (e.g., live sound)
The differences between the two woofers are rooted in their mechano-electro-acoustical interactions, which govern how electrical energy converts to mechanical motion and ultimately to acoustic output.
In essence, the differences reflect domain-specific optimizations: the second woofer prioritizes mechano-acoustical midrange precision, while the the 15NBW100 emphasizes electro-mechanical bass output.
Finally:
Apart from the phase shift inherent to mass-controlled woofers with 'flat' response, there's also an implicit efficiency loss resulting in incorrect replication of the waveshape of the input signal as byproduct. I've posted about this earlier in this thread.
Attachments
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Is this content AI generated?Lower moving mass enables faster cone acceleration (a=F/Mmsa=F/Mms), improving transient response for midrange signals.
By the way, it is not correct.
Another example regarding DACs:
"Contrary to popular belief, digital is still more complex than theoreticians/CAS/Perfect-bit people make it out to be. While digital replay standards have improved in general, all the theories and latest "advances" are often (like digital itself) merely number crunching games that get you nowhere. It still holds true that a good digital playback medium depends heavily on the ears of the designer. There are good sonic reasons why the best of the older products are still being cherished and sought after by connoisseurs."
and
"The digital signal should be handled with great care. Less processing; treat the original bitstream as fundamental truth."
In terms of preserving/respecting 'the original bitstream', this simple DAC board beats every modern (high-end) ΔΣ DAC:

Only 16bit/44.1kHz (= 100% transparent) and the distortion figures would put it at the bottom of the SINAD list.
"Contrary to popular belief, digital is still more complex than theoreticians/CAS/Perfect-bit people make it out to be. While digital replay standards have improved in general, all the theories and latest "advances" are often (like digital itself) merely number crunching games that get you nowhere. It still holds true that a good digital playback medium depends heavily on the ears of the designer. There are good sonic reasons why the best of the older products are still being cherished and sought after by connoisseurs."
and
"The digital signal should be handled with great care. Less processing; treat the original bitstream as fundamental truth."
In terms of preserving/respecting 'the original bitstream', this simple DAC board beats every modern (high-end) ΔΣ DAC:

Only 16bit/44.1kHz (= 100% transparent) and the distortion figures would put it at the bottom of the SINAD list.
Deleted.Is this content AI generated?
By the way, it is not correct.
For those who want to delve deeper into the subject, it's advisable to start with Federick Hunt's Electroacoustics.
However, some subtle details (where the proverbial devil resides) remain unexposed or underexposed even in the most advanced and extensive studies.
'Stored energy' is usually attributed solely to resonances, while there is a second (in my opinion no less important) explanation for it.
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That's very nice; thanks for sharing. The References section refers to another very good read (written by two ladies and a gentlemen, which I also find interesting), "Class D Audio Amplifier with Reduced Distortion".Both, but mainly due to the modulation of the original signal and the additional circuitry required (I've attached a paper including an overview).
- Home
- Loudspeakers
- Multi-Way
- Is it possible to cover the whole spectrum, high SPL, low distortion with a 2-way?