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Nyquist was wrong?!, interesting headwize.com article
Azeteg
post Jul 25 2003, 14:21
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QUOTE (KikeG @ Apr 27 2003, 12:05 PM)
QUOTE (jmvalin @ Apr 27 2003 - 06:01 PM)
While I agree that 44.1 kHz might not be enough (even then, I'm not completely sure), 48 kHz should be enough.

I still think that, given limitations of human hearing, 44.1 KHz is enough. Maybe for some 0.1% of young bat-eared people there could be a difference using a higher sampling rate, and just on critical material. But even in this case, the difference I believe would be very subtle. Still, I would only be convinced form a rigorous, repeatable, blind test.

Stumbling across this discussion, I just had to jump in on some of what's being discussed.


1. Filter ringing / Impulse response

Any filter, may it be analog (electronic), digital or even mechanical, will ring around fc. The steeper the filter, the longer the impulse resonse (ringing) will be. Symmetric filters (only realizable in the digital domain) will have even distribution of pre/post ringing, creating a flat phase response.


2. Audibility of filter steepness

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.


Cheers,

Martin Saleteg
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idioteque
post Jul 25 2003, 14:39
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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.
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tigre
post Jul 25 2003, 14:42
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QUOTE (Azeteg @ Jul 25 2003, 05:21 AM)
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.

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?


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Azeteg
post Jul 25 2003, 15:08
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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? smile.gif (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

This post has been edited by Azeteg: Jul 25 2003, 15:13
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tigre
post Jul 25 2003, 15:31
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QUOTE (Azeteg @ Jul 25 2003, 06:08 AM)
...
Hope this clarifies a bit?

Your explanation is clear to me. smile.gif

What I'm sceptical about:
QUOTE
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.

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

QUOTE
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.

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)?


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Azeteg
post Jul 25 2003, 15:56
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QUOTE
Sine waves at 20kHz (reasonable volume) are inaudible, so why would pre-echo/ringing of this frequency cause what you've described?


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.



QUOTE
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 guess it depends on the quality of your reproduction chain. No, a Soundblaster won't do it.
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Pio2001
post Jul 25 2003, 19:23
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QUOTE (idioteque @ Jul 25 2003, 04:39 PM)
He would tear this article to shreds.

What article are you talking about ?
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tigre
post Jul 25 2003, 19:49
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QUOTE (Azeteg @ Jul 25 2003, 06:56 AM)
I have never said anything about us being able to hear beyond 20kHz. I was never referring to steady state signals.
<snip>

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:
QUOTE
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.


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?

QUOTE
...
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.


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?

QUOTE
1. We are not talking about steady state signals.

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


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tigre
post Jul 25 2003, 20:06
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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?

This post has been edited by tigre: Jul 25 2003, 20:07


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Pio2001
post Jul 25 2003, 20:29
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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 rolleyes.gif
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F1Sushi
post Jul 25 2003, 20:51
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QUOTE (Pio2001 @ Jul 25 2003, 03:29 PM)
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  rolleyes.gif

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...
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tigre
post Jul 25 2003, 21:56
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QUOTE (F1Sushi @ Jul 25 2003, 11:51 AM)
QUOTE (Pio2001 @ Jul 25 2003, 03:29 PM)
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  rolleyes.gif

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) ... smile.gif

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 wink.gif 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. rolleyes.gif


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Pio2001
post Jul 26 2003, 01:13
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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.
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Azeteg
post Jul 26 2003, 10:18
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QUOTE (tigre @ Jul 25 2003, 10:49 AM)

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


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


QUOTE
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.


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.


QUOTE
Any neurological measurements done on this you know of?


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.

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


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
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Azeteg
post Jul 26 2003, 10:27
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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
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KikeG
post Sep 7 2003, 20:26
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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=

This post has been edited by KikeG: Sep 7 2003, 20:55
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KikeG
post Sep 8 2003, 10:08
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QUOTE (Azeteg @ Jul 25 2003, 03:08 PM)
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.

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.

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KikeG
post Sep 8 2003, 10:14
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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.
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boojum
post Sep 8 2003, 10:40
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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)


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tigre
post Sep 8 2003, 11:45
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QUOTE (KikeG @ Sep 8 2003, 01:08 AM)
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.

QUOTE
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.


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:
QUOTE
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 ...


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2Bdecided
post Sep 8 2003, 12:00
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QUOTE (tigre @ Sep 8 2003, 10:45 AM)
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.
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KikeG
post Sep 8 2003, 13:16
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QUOTE (2Bdecided @ Sep 8 2003, 12:00 PM)
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.

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.

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


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.

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tigre
post Sep 8 2003, 13:40
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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).

This post has been edited by tigre: Sep 8 2003, 13:40


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2Bdecided
post Sep 8 2003, 13:58
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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.
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tigre
post Sep 8 2003, 18:08
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QUOTE (2Bdecided @ Sep 8 2003, 04:58 AM)
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?

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).

This post has been edited by tigre: Sep 8 2003, 18:08


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