I would love to sic my college digital signals and systems professor on whoever wrote this. (His nickname was 'the hurricane') He would tear this article to shreds. Complete bullshit.

It sounds like you know what you're talking about - but this is theory. What we know about the ear (and brain) are just models, and these models are more or less close to reality but not perfect - otherwise it would be no problem samples in lossy codecs.

As you probably know it's a forum rule (#8) here that claims need to be backed up with evidence. So could you please either provide links to double blind tests or give a suggestion how to create test signals that sound different after 20kHz lowpass?

I did not know about any forum rules,

however, I have been looking into this problem ever since trying to create a close-to-perfect LowPass filter for the use in a samplerate converter. I simply sat down listening to various filter steepnesses for 96kHz->48kHz and 96kHz->44.1kHz conversions, and I noticed there were quite remarkable differences in imaging and transient response.

I have looked all over for papers describing this particular phenomenon, but to no luck. The problem is real and does exist though. It is in fact rather obvious.

Explanations (bio-digital? (Thanks to Jim Johnston for this)

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. 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.

This is an example of non-linearities introduced by tiny amounts of pre-echo.


A small test to try:

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.


A real world test to try:

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.


Hope this clarifies a bit?

Martin Saleteg
N i n j a F X

Your explanation is clear to me.

What I'm sceptical about:

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)?

I have never said anything about us being able to hear beyond 20kHz. I was never referring to steady state signals. As you probably know, transients are NOT steady state signals. And a transient can NEVER be of a single frequency alone. This would be mathematically impossible. You will find more frequencies depending on the length of the analysis window used.

So:

1. We are not talking about steady state signals.

2. We are not talking about linear systems.






I guess it depends on the quality of your reproduction chain. No, a Soundblaster won't do it.

What article are you talking about ?

I didn't say so. I'll try to clarify:

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

Your claim was:


If you lowpass a transient (e.g. silence with a single 1-sample click), pre- (and post-) ringing arround fc is introduced. Making the lowpass steeper causes
- the ringing to become louder
- the ringing to last longer
- the frequency range of ringing arround fc to become wider.
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.

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 understand your explanation about the ear very well as I'm studying medicine. It's clear and a known fact that it works like this if a pre-echo (or any "sound before a sound") is audible - but is there any proof that it works like this if the pre-ringing is in a conciously inaudible frequency range? Does the sound reach the inner hair cells at all - and, if yes, do they react? 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?

Two ideas to ABX (double blind test) your claim using ordinary equipment:

1.
a ) Take a sample with strong transients (e.g. castanetts sample or something artificial - suggestions welcome) (resample it to 48kHz if necessary because of not-so-decent soundcard, using SSRC or other HQ resampler)
b ) Apply the steepest available lowpass (e.g. CoolEdit's FFT filter) at 16, 17, 18, 19, 20, ... kHz and ABX if there's an audible difference
c ) Take the highest ABXable lowpass (e.g. 18kHz) and apply a less steep lowpass to the original arround this frequency (e.g. 17.5-18.5kHz) and
d ) try to ABX against the steep lowpassed.
e ) Result: If d ) works your claim is most likely true for the tested frequency (e.g. 18kHz), otherwise not. - Comparing to the highest frequency one can hear might be interesting.

2.
a ) See 1.a )
b ) Add a loud increasing high frequency tone (or similar e.g. narrow band noise) before transients manually
c ) ABX
d ) Result: see 1.e )

Volunteers?

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...

Good that you care about people here (and their equipment) ...

I didn't mean to create something like udial sample (high frequency 5x amplitude of audible tone) - rather like this:
- Audible signal with transient 1/2 * max. amplitude (+/- 32768)
- "Artificial pre-ringing" something like 1/10 - 1/4 * max. amplitude
- Duration of "artificial pre-ringing" (fadein) smaller than 1/2 second

I think 1. is more reallistic anyway - it's already exagerated ("Apply the steepest available lowpass" = much steeper than necessary)


I did a 1st try:
Test signal: 1-sample-clicks, ~ 10/second, created at 48kHz sampling rate
lowpassed at 20kHz using CEP FFT filter; lowpass width (100%-0%: 5.9 Hz)
ABX'ed 8/9 but then lost focus/control/luck and gave up at 9/12...
I guess I should start with something like 16kHz lowpass and increase step by step. I just thought a quick success couldn't do any harm as listening to this sample is no nice experience at all.

Don't forget to check the resulting signal at the sample level.

SoundForge 4.5 for example completely destroys the transient when applying a lowpass, it generates new transients at both ends of the impulse response, resulting in a triple transient. Its filters seem not to work properly.

I posted about this here, but it was so long ago that I don't know what to search for.

And what happens when you do this transformation? (just like the ear does)




So you agree with me, that the ringing introduced when low-passing a transient is NOT steady.

The ear analyzes this NON-STEADY ringing signal. Again, what happens when you analyze a non-steady signal with a given window length?

You will find that the ear finds energy outside of the frequency of ringing. If energy is loud and long enough, this will (as I stated before) introduce non-linearities in the ear, in other words, your perception of the transient will be different.




I have searched for papers about this for a lnog time without finding any. I know there have been some less scientific experiments done on filter steepness (one of the tests by Tom Stockham I think) and the conclusion seem to be that at fs=50kHz it is possible to create a filter that will be shorter than the ears shortest analysis window.



If a signal per definition is not steady state, it cannot be treated as such.


If you're studyiong medicine perhaps this is a very good topic for some research :-) I know a bunch of people who would be glad to have this printed black on white.


Cheers,

Martin Saleteg
N i n j a F X

Oh, I also forgot... Don't try the tests with Soundblasters, Midiman Deltas or anything in that range... The conversion process will most probably mask all of the effects.

And perform the tests at 96kHz.

Samplerate converting the source signal is not a very good way to test it either, since the samplerate converter will add an anti-imaging filter in itself. Try using 96kHz sources.

And you don't necessarily have to listen to just transients. My experiments show that stereo imaging is what is first lost when pre-echos start appearing. (Since tiny transients is what define stereo image, they get blurred -> Stereo image gets less clear)


Martin Saleteg
N i n j a F X

Azeteq: if you are still around, some blind (ABX) tests would be good to support your claims, together with the source files and some information about the characteristics of the filter used.

About pre-ringing: it is not a steady signal, but it is a narrowband signal with frequency content just around fc. I thing it is doubtful whether this signal could be heard alone, and I think it is also doubtful it having an effect in perceivability of the posterior impulse. I can think of it having an audible effect just due to nonlinearities inside the ear producing audible components. I read a post from James Johnston (JJ) about the possibility of such effect being true, but I think it could be very subtle as much. Anyway, it's something that would need to be tested.

I don't see any problems with M-Audio cards performance at 96 KHz.

We are trying to set up a blind test in order to check some of these things, see http://www.hydrogenaudio.org/forums/index....topic=12920&hl=

According to some tests I just performed, I think there is some flaw in this explanation.

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.

When someone uses statements like this: "However, using music with a substantial share of upper frequencies (soprano, hobo, upper strings) one notices that the sounds gets less brittle and that the harshness at the treble has gone." to sell something, even engineering students ought to have their BS alarm go off. The fellow who wrote the article is selling snake oil. He goes on to say that the effect is very subtle. Translation: if you do not hear the difference you are a plunk. Sounds like the emperor has new clothes to me.

This kind of prose has accompanied hustles in the audio world for all of the time I have been in it, since 1956. As a general rule, if it sounds too good too be true, it is.

L8R B)

I've done similar tests when Azeteg made his statements, no success as well.

Here 2Bdecided has summarized what Azeteg tried to say in quite understandable words:


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 ...

Maybe I misunderstood, but I think the problem with using a pure impulse to test this is the opposite to what you suggest.

A filter will only pre-/post ring when there is energy in the signal somewhere around its cut-off frequency (at least).

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".

I think it would be much better to use a very high quality recording (or a codec killer maybe - what would be a pre-ring or pre-echo killer?) and add lots of pre/post-ringing to that. That would be a good test.


I've tried it with the 2496 samples from the PCABX site, using an audiophile 2496, and HD580 headphones - no luck!

But, when I heard the advantage of 2496 over CD quality, it had nothing to do with HF response - see http://www.hydrogenaudio.org/forums/index....opic=9311&st=51

Cheers,
David.

Yes, here I was just addresing what he said about non-steady tones having a higher bandwidth using short-time analysis, and then supposedly being possibly audible.



It depends. Naoki's superEQ has pre-ringing issues that are hard to detect using actual music, but easy to detect using an impulse signal. The potentially audible mechanism here is different, but on the other side I think that music that has transients with strong content exactly at filter cutoff (somewhere from 20 KHz to 22 KHz) is not that easy to find.

Let me try to explain more understandably:

Simplified the cochlear amplifier (CA) works like an compressor, the only difference is that it can't look ahead and it needs some time to react/adjust. So if there's a signal (consciously audible or not) that triggers CA, it takes some time until the perception of the signal afterwards is changed.
So single-click transients
a ) are probably too short to change the perception of themselves by triggering CA
b ) trigger CA by their content in audible frequency range. If >20kHz content is changed e.g. by lowpassing, adding pre-ringing etc., there won't be much "extra-trigger" caused by this.

Additionally Azeteg's claim should lead to this: Not only pre-ringing introduced by lowpassing could trigger/change CA, also the loss of high frequency content due to lowpassing could do this. So a test signal should be designed like this:

Some sublte, quiet sounds and immediately before (several miliseconds) much high frequency content, most of it in inaudible range (or arround lowpass frequency).

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?

Cheers,
David.

IMO 1st we would need a proof (at least 1 successfully abxed sample) that this exists at all (or not). Testing with normal music should be a 2nd step if the 1st was successful (using samples especially created for this).

Here's a sample. My soundcard is just too crappy to be useful for testing this. high frequencies cause bumping background noise. I have to tweak the settings or try some other soundcards, I guess.

Something I just found about audibility of ultra-high frequencies:

From http://groups.google.com/groups?hl=es&lr=&...a%2Ben%2BGoogle

From an AES lecture from David Griesinger:

"Adding ultrasonics to a recording technique does NOT improve time
resolution of typical signals â?? either for imaging or precision of
tempo. The presumption that it does is based on a misunderstanding of
both information theory and human physiology.
Karou and Shogo have shown that ultrasonic harmonics of a 2kHz signal
are NOT audible in the absence of external (non-human) intermodulation
distortion.
Their experiments put a limit on the possibility that a physiological
non-linearity can make ultrasonic harmonics perceptible. They find
that such a non-linearity does not exist at ultrasonic sound pressure
levels below 80dB.
All commercial recordings tested by the author as of 6/1/03 contained
either no ultrasonic information, or ultrasonic harmonics at levels
more than 40dB below the fundamentals.
Our experiments suggest that the most important source of audible
intermodulation for ultrasonics is the electronics, not in the
transducers.
Some consumer grade equipment makes a tacit admission of the
inaudibility of frequencies above 22kHz by simply not reproducing
them. Yet the advertising for these products claims the benefits of
â??higher resolution.â?�
Even assuming ultrasonics are audible, loudspeaker directivity creates
an unusually tiny sweet spot, both horizontally and vertically."

More at http://world.std.com/~griesngr/ , at the end.

The lecture is available at http://world.std.com/~griesngr/intermod.ppt

Among other interesting things, it says and gives some examples of how most DVD-A and SACD the writer tested were just resampled from a 48 KHz master. In other words, they had no content at all over 22-24 KHz. And of the few he found had content over 48 KHz, it was in quite small amounts (40 dB below the fundamental, as much). It also shows ultrasonic noise characteristic of SACD, but somewhat surprisingly, just in one of the SACD disks tested.There are some spectrum graphs showing all this.

Also, it says that high directivity of speakers at ultrasonic frequencies, together with high attenuation of those frequencies along the way to the inner ear, makes reception of those signals possible just at a very small spot at the listening location.

There are many other explanations and test results over intermodulation, hearing internal working and such.

I know I'm attracting flamage here, but in trying to establish exactly what is "good enough", we have to remember that the process of audio-quality refinement will inevitably reach further into the realm of intangibles. There are perceptible differences, noticeable differences, and obvious differences. Obvious differences are the ones we can talk about with any degree of confidence, e.g. "wow, that 128kbit CBR file really does ring!". Noticeable differences are those we might not be able to put into words but that still affect our appreciation of the sound's quality. There are still perceptible differences that we may not be entirely conscious of, and it's this region we're starting to stray into by comparing CD-audio with DVD-A, SACD, or even the original analog signal. The sad truth is that it's almost unavoidable that differences must exist that are not justifiable within our current, limited models of the ear and its interaction with sound. The fact that serious audio enthusiasts (and I mean that in the best possible sense ) are still today talking in terms of absolute frequency/tone response when discussing human hearing shows exactly how limited our knowledge is.

We must refrain from pooh-poohing the notion that the ear is nonlinear, because to date there is simply no good reason to suppose that our models are complete. The ear is a living subsystem, and the most essential defining feature of living systems is that their responses are nonlinear!

To summarise: just because you haven't discovered the mechanism yet, it doesn't mean it 's not there.

Yes, but it means that it it doesn't affect us.
There is a difference between a fact for which we don't know the explanation, and the absence of fact ! This board's Terms of Service are strict : 8. Any statement about sound quality must be supported by the author responsible for such statements by a double blind listening test demonstrating that he can hear a difference.
This way, we have a fact to begin with, then, we can search an explanation. Without blind test results, we consider that we have no fact to explain.

For DVD-A or SACD quality, we are currently gathering facts, but they are difficult to reproduce in order to confirm them : the japanese Oohashi et al. experiment, and Listen's results.

Griesinger's experiment, that Nika Aldrich and I have confirmed, doesn't necesseraly dismiss them, because we listen for intermodulation between continuous tones, while in Oohashi et al.'s experiment, the recording used was a gamelan's recording, that is a kind of metallophone. The negative result in Griesinger's experiment is that if a 90 dB (for example) ultrasound doesn't intermodulate, then the 50 dB harmonic of a music instrument certainly won't.
But we forgot that this 50 dB measured value is an average value, while the gamelan has high frequency content mostly on attacks, thus, since the high frequency content is present only for very short times, its level must be very high in order to give a 50 dB level after averaging.
In order to get the instant level, the frequency analysis should be done with the shortest possible FFT window, so as to see how loud the ultrasounds can get, compared to the other frequencies present at the same time. As long as the FFT window is longer than the attack duration, the instant level of some high frequencies may be higher than measured, if they are shorter.