Skip to main content

Notice

Please note that most of the software linked on this forum is likely to be safe to use. If you are unsure, feel free to ask in the relevant topics, or send a private message to an administrator or moderator. To help curb the problems of false positives, or in the event that you do find actual malware, you can contribute through the article linked here.
Topic: Audibility of "typical" Digital Filters in a Hi-Fi Playback  (Read 328544 times) previous topic - next topic
0 Members and 1 Guest are viewing this topic.

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #725
Looking at some of my past posts. I'm posting the entire abstract for the ABX1950 paper to correct any possible errors and/or omissions:

http://scitation.aip.org/content/asa/journ....1121/1.1917190

"
An understanding of the over‐all process of hearing depends upon proper interpretation of the results of many individual experiments. In the field of subjective experimentation the problem has been complicated by the wide variety of test procedures that characterize available data. If a common technique could be applied to the many different types of auditory tests, such as thresholds of acuity, masking tests, difference limens, etc., the organization of these data would be facilitated. The purpose of the present paper is to describe a test procedure which has shown promise in this direction and to give descriptions of equipment which have been found helpful in minimizing the variability of the test results. The procedure, which we have called the “ABX” test, is a modification of the method of paired comparisons. An observer is presented with a time sequence of three signals for each judgment he is asked to make. During the first time interval he hears signal A, during the second, signal B, and finally signal X. His task is to indicate whether the sound heard during the X interval was more like that during the A interval or more like that during the B interval. For a threshold test, the A interval is quiet, the B interval is signal, and the X interval is either quiet or signal. For a masking test, A is the masking signal, B is the masking signal plus the signal being masked, and X is either A or B repeated. The apparatus for the ABX test is mechanized so all details of the method can be duplicated for each observer, and the variability of manual operation eliminated. The entire test is coded on teletype tape to reduce the time and effort of collecting large quantities of data.
"

For example the last sentence points out that an automated procedure was used, and it could have and probably did include a list of pre-generated unknowns that were secret to everybody involved in the test, unless someone who could see the tape (likely) got clever and started reading the punched tape which is not impossible.

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #726
Thanks for pointing this misleading data out, its another serious problem with the article. If you are going to do comparisons like this the frequency bands have to be perceptually relevant.
Can someone explain to me what MAF and UEN thresholds (in Fig.3) are and how they differ from the well known Fletcher–Munson curves ?

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #727
MAF reflects sound pressure levels in the outside world, MAP reflects sound pressure levels near the eardrum. They are both testing loudness of sinusoids. Fletcher-Munson curves are also for sinusoids, but presented via headphones, i.e. not properly accounting for filtering of the head and pinna. Also, Fletcher-Munson curves are old and less accurate than modern measurements.
UEN shows the sound pressure level of noise (not sinudoids) that appears to be uniformly loud at all frequencies (no word on where it's measured: in the free field or near the eardrum).

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #728
Good morning Arny .
Quote from: Amir link=msg=0 date=
At 3.5 Khz and eyeballing it, the noise floor of the content is around -25 db spl and the peak 90 db.  So the total range is 90+25=115.  How did you get the 70 number?


Nice job of cherry-picking from misleading data.

Boy that is a grumpy way of starting a reply .  I have now seen this phrasing used multiple times to dismiss out of hand various data point.  It must be part of the parlance of this forum.  It is another form of FUD so I hope we don't keep using the tactic when we are having an educated and deep technical discussion.  More below.

Quote
The noise spectral density is said to have been measured using a 1 Hz bandwidth, but in fact the ear hears in critical bands which are about 1000 times wider around 3.5 KHz. The well  known fallacy of measuring the perceptual qualities of music or ambient noise in narrow, constant frequency bands is being exploited.

Hmmm.  Very, very odd comment Arny.  How much a rock weighs has nothing to do with how much weight you can carry.  The signal to noise ratio of a music track is a measurement.  It has nothing to do with what part of it we can hear.  Indeed  you said it properly yourself: "The musical selection used in the Meridian tests was a good example - the noise floor was only about 70 dB below peak levels and thus easily handled with 16 bits and best practices."

See?  You made no reference to what we can hear.  Of course you didn't read the graph right so let's look at a marked up version to make it clear:

.

Your comment regarding critical band is just voodoo psychoacoustics in this context triggered by not understanding the method by which we get like data.  Stuart graphs have been normalized to make it possible to compare a peak tone to a noise level.  They need no further conversion to be used for computation of the signal to noise ratio.  Indeed the very reason for existence of that graph is to make this very point regarding test music track.

Let's put all of that aside.  We just went through JJ's slides that stipulated the ear's dynamic range at 114 db.  So we need no further mistranslation of that.  Our track has 115 db of dynamic range which means if we heard it live, we would be able to appreciate it all.  A 24-bit recording system with real 20-bit response gives us that. A 16 bit system does not.

And just to bring some variety into the discussion, here is another trusted source on this topic:

From the Professor Vanderkooy's paper which we discussed earlier on his proposal to perform this test correctly:
A Digital-Domain Listening Test for High-Resolution
John Vanderkooy
Department of Physics and Astronomy, University of Waterloo, Waterloo, Ontario, Canada N2L 3G1

Using non-subtractive flat triangular probability-function (TPDF) dither will add ?2/6 of noise power, so that the theoretical S/N ratio of 98 dB for 16-bit audio becomes about 93 dB.

While this may be audible by some listeners at elevated listening levels in a very quiet environment, especially at the beginning of a musical selection or at its final decay, its significance for audio quality is not as critical as other aspects.


This is referenced in Stuart's paper.  As you recall from our previous discussions, it outlines how one would create a much more reliable test of high resolution against down converted 16/44.1.

So I hope are done with the constant declarations that "music" is always 70 db or so and hence you don't need more than 12 to 13 bits.  That is junk audio science if you forgive me for saying so.
Amir
Retired Technology Insider
Founder, AudioScienceReview.com

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #729
Good morning Arny .
Quote from: Amir link=msg=0 date=
At 3.5 Khz and eyeballing it, the noise floor of the content is around -25 db spl and the peak 90 db.  So the total range is 90+25=115.  How did you get the 70 number?


Nice job of cherry-picking from misleading data.

Boy that is a grumpy way of starting a reply .  I have now seen this phrasing used multiple times to dismiss out of hand various data point.  It must be part of the parlance of this forum.  It is another form of FUD so I hope we don't keep using the tactic when we are having an educated and deep technical discussion.  More below.

Quote
The noise spectral density is said to have been measured using a 1 Hz bandwidth, but in fact the ear hears in critical bands which are about 1000 times wider around 3.5 KHz. The well  known fallacy of measuring the perceptual qualities of music or ambient noise in narrow, constant frequency bands is being exploited.

Hmmm.  Very, very odd comment Arny.  How much a rock weighs has nothing to do with how much weight you can carry.  The signal to noise ratio of a music track is a measurement.  It has nothing to do with what part of it we can hear.


That's right Amir, in your radical subjectivist world it matters not in the least how well measurements correlate with perception or audibility.

Amir THD and jitter with any number of leading zeroes after the decimal point is what you seem to want to sell and the records of AVS have recorded this any number of times.  How many times has this played out in your posts?

That's not the intent for the breed of objectivists of the kind I hang with.

If numbers are irrelevant to audibility, why these  papers?

http://www.gedlee.com/distortion_perception.htm


Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #730
I have a very slight concern about the filters they used, beyond the fact that they are unnecessarily steep. As far as I can tell, they didn't window them. I could be wrong on this, but let's assume for a moment I'm right. While the intended pre-/post-echos are at ~22kHz and ~24kHz, truncating the filter (rather than windowing it) can cause full-band echoes at the ends of the filter (i.e. +/-4ms). There's not enough information to know for sure, but best guess is they could be -76dB down. It could be a lot lower. It's probably not spectrally flat.

I'm not saying I think that it should be, or is, audible. I'm saying it's possibly something wrong in-band, albeit at a very low level. Temporal masking data suggests that it should be masked, though one big caveat is that temporal pre-masking declines dramatically with listener training.

It's a shame there's not that more information about the generation of the filters. They mention which MATLAB toolbox they used, but not all the parameters. A sentence more information, and those of us with MATLAB and that toolbox could re-create the exact filters ourselves.

Cheers,
David.

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #731
It is another form of FUD so I hope we don't keep using the tactic when we are having an educated and deep technical discussion.

Wow, the irony is stunning.


Quote
The noise spectral density is said to have been measured using a 1 Hz bandwidth, but in fact the ear hears in critical bands which are about 1000 times wider around 3.5 KHz. The well  known fallacy of measuring the perceptual qualities of music or ambient noise in narrow, constant frequency bands is being exploited.

Hmmm.  Very, very odd comment Arny.  How much a rock weighs has nothing to do with how much weight you can carry.  The signal to noise ratio of a music track is a measurement.  It has nothing to do with what part of it we can hear.  Indeed  you said it properly yourself: "The musical selection used in the Meridian tests was a good example - the noise floor was only about 70 dB below peak levels and thus easily handled with 16 bits and best practices."

Wow, you're obviously absolutely clueless. Do you even know what signal-to-noise ratio means?


Of course you didn't read the graph right so let's look at a marked up version to make it clear:

 
The embarrassing part here is that I already explained spectral density ~20 pages back. How resistant can a human be to learning anything?

If you thought logically, for just one minute, you'd see that by your wrong method of graphically determining "SNR" you would have to say that 16-bit RPDF results in an "SNR" of over 130 dB given that track and a "noise floor" of about -145 dB.

Anyone can spot this as patently absurd, amirm. You are also misusing the term peak SNR, like all the other "deep technical" () terms which you apparently have no clue what they actually mean.

... I can't even be bothered to decipher the rest of your nonsense.
"I hear it when I see it."

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #732
Since BS is armirs fellow he simply can ask him for the Matlab data.
Is troll-adiposity coming from feederism?
With 24bit music you can listen to silence much louder!

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #733
That's right Amir, in your radical subjectivist world it matters not in the least how well measurements correlate with perception or audibility.

To do that, you need to read the measurements right.  You were off by what, 45 db?  Not a small error.

And I quoted JJ for the audibility part.

Quote
Amir THD and jitter with any number of leading zeroes after the decimal point is what you seem to want to sell and the records of AVS have recorded this any number of times.  How many times has this played out in your posts?

THD?  The only interactions I recall are of this nature posted just last year:

Quote from: Amir link=msg=0 date=
:
THD figure is an improper metric for audibility of distortion. It does not at all follow the rules of psychoacoustics. If I have a 5 Khz signal with a second harmonic at 10 Khz vs a 1 Khz signal with second harmonic at 2 Khz at the same level, the THD will be the same (with respect to the second harmonic). Audibility of these two situations however is very different due to frequency masking and resolution of auditory filters. One must know the spectrum of the harmonic distortions to determine the audibility which of course is never given (or often not measured individually).

We use THD because it used to be easy to measure and has become so ingrained in the industry that it continues to be used. But other than a gross measure, it has little value.


Arny answers thusly: http://www.avsforum.com/forum/91-audio-the...ml#post23952991

Quote from: Arny on AVS link=msg=0 date=

THD is a reasonable metric for electronics and other situations where it can easily be reduced to the point where it is orders of magnitude below inaudible, psychoacoustics notwithstanding.

One leading critic of THD as a metric are Geddes and Lee who are good friends and whose papers I have actually read and recall a few relevant details from. Their criticism of THD is in the context of loudspeakers and other components that still may have audible nonlinear distortion.

Therefore the claim above that "...THD figure is an improper metric for audibility of distortion. " is inconsistent with science, reason and accepted recent research in the field of audio.


Clearly you are ignoring psychoacoustics and the work of Earl below in demonstrating the same in stating that.

Quote
That's not the intent for the breed of objectivists of the kind I hang with.

If numbers are irrelevant to audibility, why these  papers?

http://www.gedlee.com/distortion_perception.htm


Let's quote Earl, shall we?

This is precisely where the signal-based distortion
metrics fail. In our next paper we will show that .01%
THD of one type of nonlinear system can be
perceived as unacceptable while 10% THD in another
example is perceived as inaudible.
Even one of these
simple examples is sufficient to invalidate THD as a
viable metric for discussion of the perception of
distortion.
Furthermore, 1% THD is not at all the
same as 1% IM, but we will show that neither
correlates with subjective perception. While some of
the signal-based metrics may be “better” than others,
it is our opinion they all fall short of what we are
seeking.


I suggest instead of just having drinks with Earl, you actually listen to him and learn from him why THD is anything but "consistent with science, reason and accepted recent research in the field of audio" that  you wrote on AVS.

As JJ is fond of saying, almost any measurement that lacks a spectrum analysis, is useless in determining audibility.
Amir
Retired Technology Insider
Founder, AudioScienceReview.com

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #734
Hi Amir,

Hey buster. 

Quote
In another thread here, you linked a page with this article on your sales store website, titled "Audibility of Small Distortions" By Amir Majidimehr, the self assessed objectivist/non-hobbyist.

You exhibit a lot of frustration there partner.  Any symptoms of dry mouth and restless sleep go with that?

Quote
Quote
The “Q” indicates how steep the resonance is in the frequency domain.  In the time domain (not shown), the higher the Q, the more “ringing” the system has.  Ringing means that a transient signal (think of a spike) will create ripples that go on for some time after they disappear.  An ideal system would reproduce that transient with zero ringing.  The higher the Q of a resonance, the more ringing the system has. Reading what I just wrote, if I asked which one of the resonances on the right is more audible, you will likely say High Q.  It seems natural that it has the highest amplitude change and more time domain impact per my explanation.  Yet listening tests show the opposite to be true!  The Low Q is more audible.



Here is the measurements from Stuart's article:


What do you make of this and what would the implications be for audibility with and without typical and atypical filters?

The audibility of resonances is based on research from Dr. Toole and Dr. Olive.  It is based on listening tests of simulated resonances.  It was made at low frequencies. You can't apply it to the above graph.

For one thing, the one you have circled on the right is in the ultrasonic/inaudible region.  How the heck would you run a test and confirm Dr. Toole/Olive's results when the subjects can't hear the main tone let alone the variations from flat response?

On the prior one, the bandwidth of auditory filters are quite large by the time you get there.  See my article on that here: http://www.madronadigital.com/Library/RoomReflections.html

And this graph:



You are way at the end of that graph.  To determine audibility, you need a frequency response measurement that is psychoacoustically filtered per above.  Such a filtering would smooth out a lot of ripples there.

Quote
Is it possible for out of band resonances to create harmonics in band?

How did you jump from linear resonances to non-linear distortions?
Amir
Retired Technology Insider
Founder, AudioScienceReview.com

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #735
Wow, more quoting posts from other forums in an attempt at making others look as stupid as yourself. That's so pathetic.

amirm, how about you first learn what SNR means and come back then with a correction to your ridiculous graphical attempt at determining "SNR", as we can all see above?
"I hear it when I see it."

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #736
Wow, more quoting posts from other forums in an attempt at making others look as stupid as yourself. That's so pathetic.

??? He asked me specifically about posts on AVS: "Amir THD and jitter with any number of leading zeroes after the decimal point is what you seem to want to sell and the records of AVS have recorded this any number of times."

What is pathetic about answering his question as he asked?

That aside, do you agree with my position or his?
Amir
Retired Technology Insider
Founder, AudioScienceReview.com

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #737
Wasn't Earl's paper about speakers?

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #738
No samples arrive at the ear, as you said. The ear receives a continuous mechanical wave that oscillates. Of course the ear doesn't wait for specific filters to arrive, it would have to wait a few seconds if we use a long filter.

You don't say! 
Quote
As you've quoted many times now, the problem with pre-ringing jj mentions primarily are filter banks in codecs that operate across the whole audible range. He even specifically gives the example of cutoff frequencies of 2 and 4 kHz.

2 to 4 Khz?  Filter Banks?  Sorry no.  Not even close.  The slides could not be clearer.  Let's review them again:

.

Shouldn't have post just now.  Reminded me to correct your post above .
Amir
Retired Technology Insider
Founder, AudioScienceReview.com

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #739
Good morning Arny .
Quote from: Amir link=msg=0 date=
At 3.5 Khz and eyeballing it, the noise floor of the content is around -25 db spl and the peak 90 db.  So the total range is 90+25=115.  How did you get the 70 number?


Nice job of cherry-picking from misleading data.

Boy that is a grumpy way of starting a reply . 


Just following your lead, Amir.

Quote
Quote
The noise spectral density is said to have been measured using a 1 Hz bandwidth, but in fact the ear hears in critical bands which are about 1000 times wider around 3.5 KHz. The well  known fallacy of measuring the perceptual qualities of music or ambient noise in narrow, constant frequency bands is being exploited.

Hmmm.  Very, very odd comment Arny.  How much a rock weighs has nothing to do with how much weight you can carry.


A very odd and often very incorrect claim, especially if someone specifies their weight carrying ability in rocks or as is more common and historic: Stone.

Quote
The signal to noise ratio of a music track is a measurement.  It has nothing to do with what part of it we can hear.


I think that the above almost unbelievable claim is actually something that you believe, Amir. It shows zero insight into the fact that the purpose of audio measurements is to provide some kind of representation of the subjective experience of listening to music through  the equipment with a set of relevant objective measurements.  Believing otherwise is common among subjectivists and other poorly informed audiophiles. In fact the subjective ragazines noften teach people that measurements are always meaningless.

Quote
Indeed  you said it properly yourself: "The musical selection used in the Meridian tests was a good example - the noise floor was only about 70 dB below peak levels and thus easily handled with 16 bits and best practices."


I didn't reference that to a cherry picked measurement, or a measurement made over a 1 Hz band.  You never asked and I suspect you never knew to ask. I've been trying to explain this and related subtleties to you on numerous occasions over many months on several forums and your recent post reflects poorly on my success in that effort.

One of the key concepts of  that I've never been able to get over to you Amir is the concept of comparing comparable items. IOW, compare apples to apples.

Music and background noise are inherently non-comparable.  Music is deterministic and composed of a collection of pure tones. Random noise is not. The energy in random noise is distributed over an infinite number of different frequencies. Accurate measurements of music only require that you measure over the range of frequencies that the music occupies which is often fairly limited. Background noise measurements require measuring over far wider ranges of frequencies. Usually noise measruements are made over the full audio band or over octaves or larger fractions of an octave. Noise measurements over 1 Hz bands are generally useless.


Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #740
This is what he says about the impact of time domain response of those filters:

"If the filter has substantial energy that leads the main peak, this may be able to affect the auditory system."

So the fact that they are ultrasonic in that regard is not material because of the time domain effect.  I am afraid that torpedoes your arguments throughout this thread.


The conclusion is not adequately supported by the out-of-context quote.

The logical error that is illustrated is that of interpreting a conditional and vague  statement as if it is a general statement.

The more obvious conditional words in the quoted text are:

substantial

may

Substantial and May are not only conditional, they are also vague.

So the statement that is being used as a proof text for a global generality is both conditional (whether or not it is true in a specific case depends on unstated conditions) but it is also vague which can be illustrated by asking the questions:

How many percent or dB is Substantial?

What is the probability that is associated with May?

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #741
Wasn't Earl's paper about speakers?



No.

I linked the full text - please read it or at least the brief abstracts and if you have any further questions I would be happy to explain.

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #742
You don't say! 

But that's exactly what you said. You said individual samples arrive at the ear. Quite funny indeed. Even funnier than not even knowing what SNR is.


Quote
As you've quoted many times now, the problem with pre-ringing jj mentions primarily are filter banks in codecs that operate across the whole audible range. He even specifically gives the example of cutoff frequencies of 2 and 4 kHz.

2 to 4 Khz?  Filter Banks?  Sorry no.  Not even close.  The slides could not be clearer.  Let's review them again:

Are you blind? Are you trolling again (or should I say still)?

fund_of_hearing.ppt:
Slide 35:
“Linear Phase” (constant delay) filters do not have this phase shift, however they have a pre-ringing. In extreme cases (some older rate convertors, audio codecs) this pre-ringing is clearly audible.  Not all pre-echo is audible.

Slide 36:
- In Codecs this is a known, classic problem, and one that is hard to solve.
- In some older rate convertors, the pre-echo was quite audible.


Slide 37 shows a random filter, it says absolutely nothing about audibility. It only demonstrated the relationship between time- and frequency-domain.


adc.ppt:
Slide 63:
- The main lobe of a filter cutting off in 2.05 kHz must necessarily have a wider main lobe than the narrowest (in time) cochlear filter. df * dt >=1.
- The main lobe of a filter cutting off over 4kHz will have a main lobe a bit smaller than the narrowest cochlear filter.


Slide 64:
reuses the same random filter as above to demonstrate the relationships between domains, not audibility.
It is absolutely ludicrous to use these examples to support audibility of pre-ringing at ~40 kHz. But we know why you do this, to win a war. See amir credibility = 0.


"If the filter has substantial energy that leads the main peak, this may be able to affect the auditory system."

Yes, deliberately cherry-picking again. Yuck! And notice the may be.
Here's what is says in the next line, again:
- In Codecs this is a known, classic problem, and one that is hard to solve.


So the fact that they are ultrasonic in that regard is not material because of the time domain effect.  I am afraid that torpedoes your arguments throughout this thread.

Since you've disqualified yourself a long time ago from any reasonable and rational discourse, I enjoy your posts for their comedic value, even if they are intellectually quite dishonest. Luckily people can google this thread and see your lack of honesty.
You've just demonstrated your dishonesty again. Good job amirm.


Quote
Even a child should understand that ringing (which can be reduced or even eliminated, see previous posts) outside the hearing range is different from clearly within the hearing range.

Hopefully you see that you are completely mistaken here and you have to explain why it is hard to get, not me .

As I suspected, you do not understand what even a child would get.

I've also asked you to listen to a 4 second short test file for like ~20 times now, to shed some light on what you hear in your system.
Are you afraid to tell us?
"I hear it when I see it."

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #743
Let's quote Earl, shall we?

This is precisely where the signal-based distortion
metrics fail. In our next paper we will show that .01%
THD of one type of nonlinear system can be
perceived as unacceptable while 10% THD in another
example is perceived as inaudible.
Even one of these
simple examples is sufficient to invalidate THD as a
viable metric for discussion of the perception of
distortion.
Furthermore, 1% THD is not at all the
same as 1% IM, but we will show that neither
correlates with subjective perception. While some of
the signal-based metrics may be “better” than others,
it is our opinion they all fall short of what we are
seeking.



Actually reading the paper being cited is informative.  Please find me where in part 2 the  .01%
THD of one type of nonlinear system was shown to be perceived as unacceptable by the average listener.

Secondly its quite clear that part 2 is partially based on crossover distortion which is well known to be audible at very low percentages because of asymmetries between the measurement environment and the listening environment. It is also so rare as to be irrelevant to modern reasonable quality audio gear.

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #744
Any symptoms of dry mouth and restless sleep go with that?

Nope. No Subjectivist audiophile BS claims about anything on my website or forum escapades, about amp$, wire$, Hi-Re$, etc.
If I like how something looks, feels, etc subjectively. I make no attempt to justify it objectively, be it a Ferrari or an ML amp. I sleep quite well because of it.

The audibility of resonances is based on research from Dr. Toole and Dr. Olive.  It is based on listening tests of simulated resonances. It was made at low frequencies.

Well, you're either misinformed, lying, both, or...?

That 12k (linear distortion) hump is certainly within the audible bandwidth, even to someone with as limited hearing as yours.

For one thing, the one you have circled on the right is in the ultrasonic/inaudible region.

Right. That's where one would expect to find Beryllium tweeter breakup resonances. Now the question is do the harmonics make it down into the audio band when excited, especially when driven to 108db @ speaker, near damage territory for a 1" DR dome.
DG's tests seemed to indicate it might not be with his 2 tweeter sample. But it's not up to me or my side to do anything. It's up to your Hi-Re$ peddler camp to show that the system used in the BS test is transparent to the test and not generating false positives...and there is zero data regarding this.

How the heck would you run a test..

I wouldn't. You and the BS crew would, for any sort of honesty. Including the switching software and file alignment and....

cheers,

AJ
Loudspeaker manufacturer

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #745
I think that the above almost unbelievable claim is actually something that you believe, Amir. It shows zero insight into the fact that the purpose of audio measurements is to provide some kind of representation of the subjective experience of listening to music through  the equipment with a set of relevant objective measurements.

The purpose of my car is to be driven by me.  But my car is an object independent of me.  Measurements are measurements.  How we interpret them is different.  If you get the measurement wrong as you did, then the interpretation becomes completely false.  Which is what happened when you used the 70 db number to proceed to tell us there is 12 to 13 bits of data in there.  If you start with the right measurement value of 115 db, then you get the right bit depth which is 19.

Quote
One of the key concepts of  that I've never been able to get over to you Amir is the concept of comparing comparable items. IOW, compare apples to apples.

And that is what Stuart has done.  You see how there is only one set of axis there?  Everything is normalized to the same two axis.

Quote
Music and background noise are inherently non-comparable.  Music is deterministic and composed of a collection of pure tones. Random noise is not. The energy in random noise is distributed over an infinite number of different frequencies. Accurate measurements of music only require that you measure over the range of frequencies that the music occupies which is often fairly limited. Background noise measurements require measuring over far wider ranges of frequencies. Usually noise measruements are made over the full audio band or over octaves or larger fractions of an octave. Noise measurements over 1 Hz bands are generally useless.

Your starting point is correct.  Where you end up is completely wrong.  Stuart has literally written the book on topic of mapping noise to tone so that it can be compared to other things such as threshold of hearing that is based on test tones as was nicely explained by Alexey.  I provided my article to you on this topic explaining the same: http://www.madronadigital.com/Library/RoomDynamicRange.html.  And noted the reference at the end:

“Noise: Methods for Estimating Detectability and Threshold,” Stuart, J. Robert, JAES Volume 42 Issue 3 pp. 124-140; March 1994

Determining the "detectability and threshold" of noise requires conversion of noise power to music tones.  The paper explains how this should be done to be perceptually correct (based no ERB) and Stuart has done the same in this paper.  He provides the same reference in his most recent paper:

[30] J. R. Stuart. Noise: methods for estimating detectability and threshold. AES 93rd Convention, Berlin, 1993.

And this is where it is used:

This [playback] level was chosen for comfort, and because it was high enough for details to be audible but also low enough that 16-bit RPDF dither would be inaudible at the listening position [30].

RPDF dither is noise.  And for it to be plotted correctly to show inaudibility requires that it be converted to the threshold of hearing base on tones. By making sure 1 Hz bins are used for the FFT of the music, we know there is no process gain and hence the numbers can be compared.
Amir
Retired Technology Insider
Founder, AudioScienceReview.com

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #746
Any symptoms of dry mouth and restless sleep go with that?

Nope. No Subjectivist audiophile BS claims about anything on my website or forum escapades, about amp$, wire$, Hi-Re$, etc.
If I like how something looks, feels, etc subjectively. I make no attempt to justify it objectively, be it a Ferrari or an ML amp. I sleep quite well because of it.

Seeing how you go to shows and exhibit all the symptoms of subjectivism with modded players, ribbon speaker cables, etc, I say your are in denial and hence the reason I asked about other symptoms. 

Quote
The audibility of resonances is based on research from Dr. Toole and Dr. Olive.  It is based on listening tests of simulated resonances. It was made at low frequencies.

Well, you're either misinformed, lying, both, or...?
Here is my graph in my article you asked about:



And here is the reference in Dr. Toole's Book:



This is the text for it:

Figure 19.9 shows examples of deviations from fl at for high- (50), medium-
(10), and low- (1) Q resonances at three frequencies when they were adjusted
to the audible threshold levels using pink noise in an anechoic chamber and
for the 200 Hz resonances detected when listening to typical close-miked,
low-reverberation pop and jazz.


So the only data for music is at 200 Hz which is what I referenced.  The rest is for pink noise but even that was tested at max of 5 Khz, not the frequencies you asked me about.

The graph you are showing is a threshold shift between anechoic and reverberant space.  That is a differential score for a different type of test.

Quote
That 12k (linear distortion) hump is certainly within the audible bandwidth, even to someone with as limited hearing as yours.

I don't know Ammar.  Maybe I need the snake oil flat ribbon speaker cable you use with your speakers to hear them.

That aside, the hump in there playing the exact same thing in all the tests at 12 Khz.  In other words it is invariant to the test.  We are not performing a speaker review here Ammar.
Amir
Retired Technology Insider
Founder, AudioScienceReview.com

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #747
For one thing, the one you have circled on the right is in the ultrasonic/inaudible region.

Right. That's where one would expect to find Beryllium tweeter breakup resonances.

Expecting and finding are two different things.  Here is the explanation again:



The *peak* level there is about 0 dbfs (relative to 105 db fs).  The floor is about -30 db fs.  You are saying that a Beryllium tweeter breaks up playing such low levels?

But let's say it does.  Go ahead and tell us what the level of the IM sidebands would be.

Quote
DG's tests seemed to indicate it might not be with his 2 tweeter sample. But it's not up to me or my side to do anything. It's up to your Hi-Re$ peddler camp to show that the system used in the BS test is transparent to the test and not generating false positives...and there is zero data regarding this.

And show we have.  The Stuart paper has won an award for best peer reviewed paper at this year's conference.  No one is waiting for lay people with no industry experience to approve anything.  No medical research waits for approval of patients.  Somehow you are confused thinking audio field is an exception because you have two ears.  You need to fill the space between them with audio knowledge which doesn't happen by reading forum posts.

That is easy to show with your own reference. This is what David Griesinger, the author in that powerpoint says about Kiryu and Ashihara’s listening tests:

Their choice of source signal MAXIMIZES the (possible) audibility of an ultrasonic signal.
The sound pressure of the ultrasonic harmonics are equal to the sound pressure of the harmonics below 20kHz.


Now look at the ultrasonic components in Stuart's test above.  There is no way any harmonic distortion in the ultrasonic range in stuart's test would create power that is equiv. to in-band frequencies.
Amir
Retired Technology Insider
Founder, AudioScienceReview.com

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #748
If you get the measurement wrong as you did, then the interpretation becomes completely false.  Which is what happened when you used the 70 db number to proceed to tell us there is 12 to 13 bits of data in there.  If you start with the right measurement value of 115 db, then you get the right bit depth which is 19.

Words from a man that has posted this image:


This shows that you not only got the measurement wrong, you don't even know what the measurement is let alone what you measured.

I just analyzed the whole Haydn track. RMS amplitude of the noise floor at the end of the track is -58 dB +/-10 dB of that (min/max using a 50ms window).
With a max RMS amplitude of the total track reaching -10 dB, the dynamic range is about 60 dB, about 70 dB if you want to use 0 dBFS as peak.

If we filter out the noise below ~300 Hz we still get a -70 dB RMS noise floor.
(This is all relative to 0 dBFS, obviously.)


And that is what Stuart has done.  You see how there is only one set of axis there?  Everything is normalized to the same two axis.

No, you still don't understand.
You cannot subtract some musical peak amplitude from the noise spectral density, which is precisely what you did to calculate "SNR" (well, you thought you did, but you were not even close).


"This [playback] level was chosen for comfort, and because it was high enough for details to be audible but also low enough that 16-bit RPDF dither would be inaudible at the listening position [30]."

RPDF dither is noise.  And for it to be plotted correctly to show inaudibility requires that it be converted to the threshold of hearing base on tones. By making sure 1 Hz bins are used for the FFT of the music, we know there is no process gain and hence the numbers can be compared.

It seems that you have no clue what you're talking about.

Yes, dither is noise.
This noise was however not converted in any way in the plot. Threshold of audibility of noise is shown as another line, which itself is unrelated to the noise floors.
1 Hz bins in the FFT does not ensure that there is no "process gain" (whatever that is in your mind) and the peak levels can actually be plotted accurately with a much higher bin width.

Let me repeat from my previous post:
If you had thought logically for just one minute you'd have seen that by your wrong method of graphically determining "SNR" you would have to say that 16-bit RPDF results in an "SNR" of over 130 dB given that track and a "noise floor" of about -145 dB.


You really should look up some basic literature about signal analysis.
"I hear it when I see it."

Audibility of "typical" Digital Filters in a Hi-Fi Playback

Reply #749
The *peak* level there is about 0 dbfs (relative to 105 db fs).  The floor is about -30 db fs.  You are saying that a Beryllium tweeter breaks up playing such low levels?

You seem confused. 105 dB SPL not FS. Also, the noise floor is not at -30 dBFS, not even close.

The interesting questions are:
What is the crossover frequency for the tweeter?
How much power does it receive during loud passages?
Where are the measurements for this tweeter?


And show we have.  The Stuart paper has won an award for best peer reviewed paper at this year's conference.  No one is waiting for lay people with no industry experience to approve anything.  No medical research waits for approval of patients.  Somehow you are confused thinking audio field is an exception because you have two ears.  You need to fill the space between them with audio knowledge which doesn't happen by reading forum posts.

Search this post for "homeopathy".
A paper with 60% positive outcome, peer-reviewed or not, doesn't make something a fact.

Peer-reviewed papers are not guaranteed to be correct either. You'd have to accept all kinds of nonsense if this was your "standard" (well, if it fits your agenda there doesn't seem to be much of a standard).


(Btw, I am not asserting that the tweeter caused audible artifacts, in case you want to attack some straw man.)
"I hear it when I see it."