As it has been stated in this thread, it should be impossible to hear filter ringing with fc beyond human hearing. This is INCORRECT. It might be tempting to think so. The human hearing cannot hear steady-state sines over 20kHz. This has been concluded in millions of hearing tests all over the world. However, when listening to impulse responses, we have to take into account what is analyzing these sounds. The human ear has its own set of filters, analysis windows. Analyze a long enough impulse response with an auditory model and you will find that these filters are indeed triggered. This is why we can hear steepness of filters even when fs=96kHz and fc=47kHz.

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Hope this clarifies a bit?

Consider a pre-echo of an impulse that is longer than the leading edge of the cochlear filter. This pre-echo will get the inner hair cells into detection mode.

Create a windowed sinc FIR filter with a narrow transtition band, place fc at 22kHz. Use a very good 96kHz recording, preferrably surround or stereo, playing back through a set of very good converters and speakers. Then gradually increase steepness of filter (or easier, apply the filter several times) until you can hear a difference.

Sine waves at 20kHz (reasonable volume) are inaudible, so why would pre-echo/ringing of this frequency cause what you've described?

I guess high resolution equipment (D/A convertor) is needed for this. - Do you think there's a way to test this with "ordinary" 48kHz soundcard (+ good headphones)?

He would tear this article to shreds.

I have never said anything about us being able to hear beyond 20kHz. I was never referring to steady state signals.
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As it has been stated in this thread, it should be impossible to hear filter ringing with fc beyond human hearing. This is INCORRECT. It might be tempting to think so. The human hearing cannot hear steady-state sines over 20kHz.

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This pre-echo will get the inner hair cells into detection mode. This starts the outer hair cells depolarizing and going into compression mode (by detuning basilar vs. tectoral membranes). When the center of signal arrives (a transient) instead of the full level, the compression has reduced the sensitivity of the system, so the central impulse does not sound as loud.

1. We are not talking about steady state signals.

For god's sake, don't try 2b on speakers unless you really know what you're doing !

Udial.wav have already fried some tweeters around here 

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For god's sake, don't try 2b on speakers unless you really know what you're doing !

Udial.wav have already fried some tweeters around here 

This is sound (no pun intended) advice. I fried a pair of tweeters in my PSBs while doing some high frequency testing about 6 months ago (Madisound gets honorable mention here for excellent and affordable replacement Vifa tweeters). The rule of thumb here is that the volume control is a dangerous weapon to your tweeters when auditioning full-scale sinewaves in the upper end of the audio spectrum.

Just because you can barely hear a high frequency tone does not mean that you aren't sending tweeter coil frying energy through your speaker cables. Exercise extreme caution with this kind of testing...

I know that transients don't consist of one frequency, nevertheless they can be transformed to frequency domain (and back).

So why should adding a 19.5-20.5 kHz increasing + decreasing sound to a signal change the signal noticably if we can't hear the added sound itself at all? I've tried this with CoolEditPro. Even if silence with a single 1-sample click is lowpassed (using fft filter) at 20kHz with a lowpass width of only 6 Hz, the frequency range of ringing is between 19500 and 20500 Hz, so still out of audible range.

Any neurological measurements done on this you know of?

Well, pre-/post-ringing is not stead state, but short "steady state with fadein and fadeout". Why should this cause a difference in perception?

When you (like the ear does) analyze a very short section of sound (say an impulse with filter ringing around 22kHz) the signal IS NOT as narrowband as you might think. Try making a short-term Fourier Transform of a sliding window on a standard AA or AI filter. Use the fastest cochlear filter length, about 400us, as the window length.

I have lowpassed a 96 KHz impulse with various ringing length passband FIR filters, leaving just frequencies between 19 KHz and 21 KHz, so that the result consists just of time-enveloped ringing at cutoff frequencies. I can't hear anything when listening to this ringing. Even when it is a non-steady signal, it does not have audible components.

I think we can't hear over 20 KHz, be it with steady signals or transient signals.

I couldn't use good equipment for this test, but still I can hear 16 KHz tones clearly with it, and up to 18 KHz, but very softly. I'll repeat this test with better equipment when I have time.

I also tried to ABX a 96 KHz impulse signal filtered with a long ringing 20 KHz lowpass FIR filter, from the unfiltered version. I couldn't. I used good equipment this time. This suggests that the effect, if existing, is very subtle as much.

If I understand him correctly, the idea is that the cochlear amplifier (ie. the active process within the cochlea, which isn't fully understood I hasten to add!) does respond to HF sound that we can't actually hear when presented as a steady state tone. This response isn't to let us hear HF sound, but to trigger a change in the cochlea tuning and dynamic compression so that audible sounds are perceived differently.

So to test this, single-sample clicks probably won't help, as they contain much audible content that will trigger the cochlea tuning anyway. I'm trying to create some samples to test this right now ...

We're trying to test if this ringing has an audible effect on the perception of other frequencies. So, we need those other frequencies to be present. i.e. using just 18-22kHz info is no good, because there's nothing else (very) audible for it to have an impact on.

Using just an impulse isn't very good either, because it's an uninteresting signal with which to judge if the sound has "changed".

But looking for strong or loud sounds around the cut-off frequency is surely wrong - it's good enough that there's some sounds around this frequency - as is typical of most music.


Surely the hypothesis isn't that "certain signals with sharp transients and/or lots of ultrasonic information are affected by this", it's that "most music is affected by this". That's the bold claim that seems to have been made - isn't that what we should test?