IPB

Welcome Guest ( Log In | Register )

7 Pages V  « < 5 6 7  
Reply to this topicStart new topic
'Normalization' of PCM audio - subjectively benign?
krabapple
post Sep 12 2006, 19:44
Post #151





Group: Members
Posts: 2519
Joined: 18-December 03
Member No.: 10538



QUOTE (Axon @ Sep 8 2006, 16:56) *
I'll take on the dynamic range article if somebody else comments on the sampling article.

EDIT: I replied to that thread, comments on my analysis welcome.



Thanks Axon, perhaps Clint will consider replacing her article with yours? I suspec though that he'd want some actual data from using your protocol.

This post has been edited by krabapple: Sep 14 2006, 07:24
Go to the top of the page
+Quote Post
Radetzky
post Sep 21 2006, 13:48
Post #152





Group: Members
Posts: 95
Joined: 10-March 03
Member No.: 5399



QUOTE (pepoluan @ Sep 8 2006, 10:23) *
Yes, all majors in EE requires a knowledge of Fourier transforms.

However I finally recall that Discrete Signal Processing (in my university) is mandatory only for the Telecommunications major. It is optional for all the other majors... and due to its complexity, naturally no one wants to take that satanic course huh.gif


Where I did my computer engineering course (University of Sherbrooke in Quebec, Canada), all Computer Engineering & Electrical Engineering students had to take a "Communications" course where we would be exposed to Fourrier and more generally to Laplace transforms. We would study Shanon and Nyquist. The final project was the simulate, using Matlab, the quantization & sampling process occuring with music @ 16/44.1. We had to reconstruct the signal and show the difference with the original signal. How cool is that? wink.gif It was very instructive.
Go to the top of the page
+Quote Post
krabapple
post Jul 29 2008, 20:21
Post #153





Group: Members
Posts: 2519
Joined: 18-December 03
Member No.: 10538



QUOTE (krabapple @ Sep 8 2006, 15:08) *
I posted a complaint thread about Tham's stuff on Audioholics. Any takers here for the reply from the editor?

http://forums.audioholics.com/forums/showt...6095#post206095


QUOTE
Perhaps you can direct the more knowledgable people on those forums to construct a new test sequence that is more real world? We're open to additional input.
__________________
Clint DeBoer
Editor in Chief
Audioholics




Something (a glitch?) made that thread appear invalid for awhile. But it's still there.

http://forums.audioholics.com/forums/showthread.php?p=206095

This post has been edited by krabapple: Jul 29 2008, 20:25
Go to the top of the page
+Quote Post
pdq
post Jul 29 2008, 20:54
Post #154





Group: Members
Posts: 3450
Joined: 1-September 05
From: SE Pennsylvania
Member No.: 24233



The link doesn't work for me.
Go to the top of the page
+Quote Post
Axon
post Jul 29 2008, 21:24
Post #155





Group: Members (Donating)
Posts: 1985
Joined: 4-January 04
From: Austin, TX
Member No.: 10933






I can't see the article either.

EDIT: I will save a "J'ACCUSE!" moment for a short while longer. This could get sensitive.

This post has been edited by Axon: Jul 29 2008, 21:28
Go to the top of the page
+Quote Post
krabapple
post Jul 30 2008, 05:10
Post #156





Group: Members
Posts: 2519
Joined: 18-December 03
Member No.: 10538



QUOTE (Axon @ Jul 29 2008, 16:24) *



I can't see the article either.

EDIT: I will save a "J'ACCUSE!" moment for a short while longer. This could get sensitive.



Yes indeed, it appears we have a problem, Houston.

Well, I learn something new every day. wink.gif
Go to the top of the page
+Quote Post
Axon
post Jul 30 2008, 05:16
Post #157





Group: Members (Donating)
Posts: 1985
Joined: 4-January 04
From: Austin, TX
Member No.: 10933



Yeah... it's gone "sensitive".
Go to the top of the page
+Quote Post
krabapple
post Aug 1 2008, 17:09
Post #158





Group: Members
Posts: 2519
Joined: 18-December 03
Member No.: 10538



QUOTE (Axon @ Jul 30 2008, 00:16) *
Yeah... it's gone "sensitive".


and now....it's back.
Go to the top of the page
+Quote Post
SpasV
post Aug 3 2008, 23:57
Post #159





Group: Banned
Posts: 69
Joined: 20-July 07
Member No.: 45497



smile.gif It is obvious you know that, unless you normalize at the pick amplitude, the normalization changes the sound. Another question is how audible it is.
If normalization decreases the sample values it introduces spectrum distortions also.
In any case, I think, it does not influence the quantization noise as seems to me the discussion is about.

For the simple case when the quantization noise is considered “white”, and this model is realistic, as it is said in “Discrete Time Signal Processing” by Alan V. Oppenheim and Ronald W. Shafer when “…the signal is sufficiently complex and quantization steps are sufficiently small so that the amplitude of the signal is likely to traverse many steps from sample to sample”,
I think: The quantization noise depends only on the quantizer used and does not depend on the signal.

So, to me, the normalization in the case of increasing the signal does not distort it as long as it stays within the acceptable number range.

This post has been edited by SpasV: Aug 4 2008, 07:27
Go to the top of the page
+Quote Post
krabapple
post Sep 7 2008, 17:04
Post #160





Group: Members
Posts: 2519
Joined: 18-December 03
Member No.: 10538



Once more into the breach....

Over on AVSForum, someone named 'DulcetTones" cited the 'Mother of Tone' page as an authoritative reference against the idea that we get 'perfect sine waves' from digital audio,

http://www.avsforum.com/avs-vb/showpost.ph...mp;postcount=74


QUOTE
I might as well point to these again as some seem intent on saying the usual vague facts.
http://www.mother-of-tone.com/cd.htm
Article is correct, no-one showed anything wrong last time. Apart from some vague comments as usual.
In other words those who say you end up with a perfect sine wave are wrong.
It seems for some reason there are a few who decide to talk about CD and say its a sinewave and yet do not understand the significance of the rectangular sinusoidal component.
This is shown in the above link, you only end up with the sine wave after the reconstruction filters work their logic on the rectangular sinusoidal components shown in the URL.
Also the URL is good as it shows some other challenges with sine waves and PCM.


I pointed out that that article had been, um, *critiqued* on HA.org, (pointing to this thread) as being rather skewed.
(I also agreed that we don't get 'perfect' reconstruction of the input signal from any audio repro medium, but digital properly done is the closest by far)

Much hooing and hahing ensues, during which DT claims HA folk don't know what they're talking about. Finally this:

http://www.avsforum.com/avs-vb/showpost.ph...p;postcount=120

QUOTE
Krab,
Ok to prove which of the two sources we are quoting actually know what they are talking about.
The person you hold in high regard who is basically stating Mother of Tone do not know what they are talking about commented in the forum you hold in high regard that Mother got the Nyquist and Shannon formulas wrong way round.
In fact your guy was quite explicit in saying this was a fundamental mistake.

Well hate to burst your bubble bud:
1st Part of the equation discussion on Mother.
QUOTE

Well, what did Nyquist say ?

maximum data rate in a noiseless channel = C = 2*W log base2( L ) bits/sec

* where 2W is 2 times the highest frequency contained in the noiseless channel, and

* where L = number of discrete levels (e.g., binary = two levels, 0 and 1)

As Nyquist seems to have been more interested in data transmission than in high-fidelity, we should not wonder, that his statement just defines a maxiumum data-rate of a communications channel.


This is EXACTLY CORRECT and pertains to the original Nyquist paper I suggested you read.
http://www.nctt.org/pages/resources/simulations/nyquist.php

Nice simple comparison, exact formula.

Then the next part in Mother expanding on the original works of Shannon:

QUOTE
Later, Claude Shannon said:

If a function s(x) has a Fourier transform F[s(x)] = S(f) = 0 for |f| > W, then it is completely determined by giving the value of the function at a series of points spaced 1/(2W) apart. The values sn = s(n/(2W)) are called the samples of s(x).

This goes much further than Nyquist's words, in that it states, that a signal which consists of sine waves with a maximum frequency of W is completely described by recording its values twice as fast as W.

The real cool thing is that Shannon also gave a interpolation formula to get back to the original signal:


Unfortunately this formula includes an infinite sum... What does that mean ?


Again this can be seen from a link that just focuses on the data from Nyquist and Shannon.
http://www.fact-index.com/n/ny/nyquist_sha...ng_theorem.html

Save the effort of reading it all here it is:

QUOTE
If a function s(x) has a Fourier transform F[s(x)] = S(f) = 0 for |f| > W, then it is completely determined by giving the value of the function at a series of points spaced 1/(2W) apart. The values sn = s(n/(2W)) are called the samples of s(x).
Also the Mother of Tone went on to show the Interpolation formula (that does not copy into here) and this matches again EXACTLY the paper by Shannon (can find it in Section II The Sampling Theorem).


So the guy your quoting actually made what I would call FUBAR statement in his argument against Mother of Tone.

The person you quote argues just as persistently as you do, makes mistakes such as stating the "infinite filter" are already here due to filters having 10000s of taps, this is rather obtuse and why I asked you earlier do you know which products are using 256 or 1024 tap lengths.

Notice 1024 is substantially smaller than the 10000s your colleague states and the 1024 type are only found in the significantly more expensive DAC-CD players.

Now why the heck would I want to wade into the bull on another forum when it is giving me a headache arguing with someone even on this forum who does not even take the time to read and follow up on both :
Nyquist Certain Topics in Telegraph Transmission Theory
Shannon Communication in the Presence of Noise



(NB: DT appears to have conflated what MikeG wrote about 'Mother of Tone' getting Shannon and Nyquist crossed, with what 2bdecided said about 'several thousands' (not '10000s') of taps....conflating at least two people, or possibly three if SebastianG's objections are included, into one)

Anyway, I suggested that rather than me arguing by proxy for those on HA that he considers incompetent and wrong about the 'Mother of Tone' article, that he address you directly here, but as you see he demurs. So I'm bringin' it here, for said parties to respond ...or not.

Btw, I of course have no issue with either Shannon or Nyquist, and my issue on that AVSF thread was with people pointing to the dreaded 'stairsteps' views as an indication of how supposedly 'problematic' Redbook is at various frequencies, and also with the lack of audibility test data from people who expect DACs that measure differently, to routinely sound different.

EDIT: fixed broken URLs in the quote

This post has been edited by krabapple: Sep 11 2008, 20:15
Go to the top of the page
+Quote Post
SebastianG
post Sep 8 2008, 09:36
Post #161





Group: Developer
Posts: 1318
Joined: 20-March 04
From: Göttingen (DE)
Member No.: 12875



As I understand it DT acknowledges the sampling theorem and goes on to say that ideal reconstruction is impossible because the ideal reconstruction filter's impulse response (normalized sinc curve) extends infinitly to both sides. So far, so good. But in practice this is not an issue. You can always design a practical reconstruction filter that is good enough. Take CDDA for example: it's not a problem to design a practical filter that faithfully reconstructs the audible band (0-20 kHz) and rejects image frequencies (those above 22.05 kHz) by 100 dB or more. The impulse response of a filter like this can be as short as 2 milliseconds:


QUOTE
In other words those who say you end up with a perfect sine wave are wrong. [...] significance of the rectangular sinusoidal component [...]

Considering practical implementations of reconstruction filters the first part is -- strictly speaking -- correct and the second part is an indication of lack of understanding. Regarding the first part: You can get arbitrarily close to that ideal with real filters. This is really just splitting hairs and only serves to confuse people.

The stair step signals shown at "mother-of-tone" is just a bad visualization of the waveform. But yeah, some stair step signal is usually an intermediate signal in current DACs that exists after digitally resampling to a much higher sampling rate and prior to analogue filtering. In a reconstruction process you're supposed to convolve a train of differently scaled dirac pulses with the reconstruction filter's impulse response. Is the stair step thingy an issue? No. Why not? Because the stair step signal is what you get after convolving the dirac pulse train with a rectangular window function. The effect of this is known (attenuation of 3.9dB at fs/2) and can be compensated for. If you do 8X oversampling or more you don't even have to worry about this because the expected attenuation of 20 kHz would be 0.046 dB or less.

Bottom line: PCM isn't flawed. You don't need ideal brickwall lowpass filters. Stair step images are misleading. "Beating" is not aliasing and is not present in the sampled data but the effect of some reconstruction filters ... the type of reconstruction filter that doesn't feature strong rejection of image frequencies near the Nyquist frequency. There's nothing wrong with this per se especially since those image frequencies may be well above the audible spectrum which is the case with CDDA.

QUOTE
SebastianG does not know what he is talking about if he states stair step and does not even use the correct name or actually know where the sinusoidal rectangular component resides in the process, I cannot take him seriously.

Oh wow. I admit that I made up the term "stair step signal". I hope it's clear what I meant, though: A series of differently scaled dirac pulses convolved with rectangular window where window width = pulse spacing. But what the heck is a "sinusodial rectangular component"? tongue.gif

Cheers,
SG

This post has been edited by SebastianG: Sep 8 2008, 18:32
Go to the top of the page
+Quote Post
krabapple
post Jul 15 2009, 19:13
Post #162





Group: Members
Posts: 2519
Joined: 18-December 03
Member No.: 10538



A heads up:

uart's 'scope screen cap on this thread. has been referenced (negatively) in a reply by Dr. Milind Kunchur to critiques (from Woodinville on Stereophile's forum, and, apparently, from other critics on other forums) of his published claims about the insufficient 'time resolution' of Redbook, briefly summarized here

http://www.physics.sc.edu/kunchur/Acoustics-papers.htm

Woodinville's (and Axon's) initial and new critiques are here

http://forum.stereophile.com/forum/showfla...age=0#Post69464

Dr. Kunchur's reply is here:

http://www.physics.sc.edu/kunchur/papers/FAQs.pdf

This post has been edited by krabapple: Jul 15 2009, 19:15
Go to the top of the page
+Quote Post
Woodinville
post Jul 16 2009, 00:47
Post #163





Group: Members
Posts: 1414
Joined: 9-January 05
From: In the kitchen
Member No.: 18957



QUOTE (krabapple @ Jul 15 2009, 11:13) *
A heads up:

uart's 'scope screen cap on this thread. has been referenced (negatively) in a reply by Dr. Milind Kunchur to critiques (from Woodinville on Stereophile's forum, and, apparently, from other critics on other forums) of his published claims about the insufficient 'time resolution' of Redbook, briefly summarized here

http://www.physics.sc.edu/kunchur/Acoustics-papers.htm

Woodinville's (and Axon's) initial and new critiques are here

http://forum.stereophile.com/forum/showfla...age=0#Post69464

Dr. Kunchur's reply is here:

http://www.physics.sc.edu/kunchur/papers/FAQs.pdf


The equivocation and attempt to deflect is really pretty astonishing to me.

The unwillingness to actually address the facts of the frequency response difference implied by a two-tap FIR filter with the two taps 5 microseconds apart is, for me, the final straw. Referring to a mechanism that creates an audible difference within the 20kHz band as mathematical nitpicking is simply not a reasonable response.


--------------------
-----
J. D. (jj) Johnston
Go to the top of the page
+Quote Post
uart
post Jul 16 2009, 02:16
Post #164





Group: Members
Posts: 810
Joined: 23-November 04
Member No.: 18295



QUOTE (krabapple @ Jul 15 2009, 11:13) *
A heads up:

uart's 'scope screen cap on this thread. has been referenced (negatively) in a reply by Dr. Milind Kunchur to critiques (from Woodinville on Stereophile's forum, and, apparently, from other critics on other forums) of his published claims about the insufficient 'time resolution' of Redbook, briefly summarized here

http://www.physics.sc.edu/kunchur/Acoustics-papers.htm

Woodinville's (and Axon's) initial and new critiques are here

http://forum.stereophile.com/forum/showfla...age=0#Post69464

Dr. Kunchur's reply is here:

http://www.physics.sc.edu/kunchur/papers/FAQs.pdf


That's interesting, I hadn't seen that pdf-file reply before. Just one quick question, in the reply he repeatedly makes reference to the "5 microsecond or better temporal resolution of the human hearing". Does anyone know where he got that figure from or how that was measured? It seems a bit extreme to me.
Go to the top of the page
+Quote Post
Axon
post Jul 16 2009, 04:55
Post #165





Group: Members (Donating)
Posts: 1985
Joined: 4-January 04
From: Austin, TX
Member No.: 10933



He measured it himself, in some (peer-reviewed!) papers, and actually, it's a figure that nobody involved (jj or krab or I) really disagree with. What this whole spat is about is the meaning of such a result, especially in the context of the fidelity of redbook, and in a larger context, the meaning of the term "resolution".

There's a really long backstory/flamewar behind all of this that you have not been introduced to (and I won't even try to do so). krab only posted because I used your scope shots as a trivial demonstration of the ability of mainstream sound cards to reproduce 14khz sine waves without distortion. Kunchur's response to your plot is a little bizarre to me, but I'm still crunching on everything before I will comment any further.
Go to the top of the page
+Quote Post
Woodinville
post Jul 16 2009, 21:11
Post #166





Group: Members
Posts: 1414
Joined: 9-January 05
From: In the kitchen
Member No.: 18957



QUOTE (Axon @ Jul 15 2009, 20:55) *
He measured it himself, in some (peer-reviewed!) papers, and actually, it's a figure that nobody involved (jj or krab or I) really disagree with. What this whole spat is about is the meaning of such a result, especially in the context of the fidelity of redbook, and in a larger context, the meaning of the term "resolution".

There's a really long backstory/flamewar behind all of this that you have not been introduced to (and I won't even try to do so). krab only posted because I used your scope shots as a trivial demonstration of the ability of mainstream sound cards to reproduce 14khz sine waves without distortion. Kunchur's response to your plot is a little bizarre to me, but I'm still crunching on everything before I will comment any further.


Specifically, he plays one pulse, and then two half-amplitude pulses separated by 5 microseconds, and listeners with unimpaired hearing in good settings can hear the difference.

Given that this is a comb filter with its first zero at 100kHz, being able to hear the difference at 20kHz isn't too surprising, now.

So, this shows a mechanism for which purely in-band (i.e. dc - 20kHz) signal can allow the distinction.

Let's see....

Down about .45dB is just at the DL. In really good conditions, for direct, immediate comparison, with no level roving, etc.

So, what did the doctor actually discover?

Nothing that I can tell.

This, of course, after misdirection upon misdirection from the people 'defending' his work, starting with quotes that made things look like it was interaural resolution (which is also audible, just barely, in the best conditions, at that level), defending his claim that you couldn't get that kind of time resolution out of a PCM signal, rejecting plots showing otherwise, changing the claims and the goalposts repeatedly and at the same time accusing the people trying to have a dialog of "shifting their excuses", and repeated, exhaustive references to Dr. K's PhD and how publishing this in a conference record was the full endorsement of the organization, etc.

The people who are saying this do, I think, know better, too.

This post has been edited by Woodinville: Jul 16 2009, 21:12


--------------------
-----
J. D. (jj) Johnston
Go to the top of the page
+Quote Post
Woodinville
post Jul 17 2009, 04:43
Post #167





Group: Members
Posts: 1414
Joined: 9-January 05
From: In the kitchen
Member No.: 18957



Oh, and the best joke of all this was that the intentionally ignorant over there think I was being rude.

Oh, baby, they have no idea what RUDE is. Apparently Gauss, Fourier, Fletcher and Zwicker are also rude, in any case.

It's just like creationists, they only accept confirmatory data.


--------------------
-----
J. D. (jj) Johnston
Go to the top of the page
+Quote Post
2Bdecided
post Jul 17 2009, 10:10
Post #168


ReplayGain developer


Group: Developer
Posts: 5364
Joined: 5-November 01
From: Yorkshire, UK
Member No.: 409



It would be fun to repeat the experiment, but hook a CD recorder into the loop, testing if (proving that!) the "effect" was still audible after passing through 44.1kHz / 16-bits.

Hopefully that would be simple enough to show how silly this is.

Cheers,
David.
Go to the top of the page
+Quote Post
instaud
post Jul 17 2009, 13:56
Post #169





Group: Members
Posts: 7
Joined: 14-July 09
Member No.: 71459



QUOTE (2Bdecided @ Jul 17 2009, 11:10) *
It would be fun to repeat the experiment, but hook a CD recorder into the loop, testing if (proving that!) the "effect" was still audible after passing through 44.1kHz / 16-bits.

Not only fun, but essential to prove his claim that CDDA is not good enough, and furthermore a test that his recommended 192kHz is the way to go. Without that, where is the relation between his artificial laboratory tests (5 years long and so very PhD-like theoretical), and reality of music reproduction?

Prof. Kunchur talks about the big picture which his results have to be viewed within. He himself writes in his FAQ about errors that are acceptable in consumer audio, but not in his experiment. So even if he found an error in CDDA, is it reason enough to replace it? He says he is an audiophile, and that's the reason his answer is yes. I come to this conclusion because he has "experienced hearing loss ... from riding sports motorcycles", listens to cassettes and LPs, but on the other hand was not happy with the sound of his CD transport/DAC combination, so he needed a "jitter buster" appliance. So the big picture for me is biased research which is funded by those having an interest in a certain result.
Go to the top of the page
+Quote Post
krabapple
post Jul 17 2009, 15:03
Post #170





Group: Members
Posts: 2519
Joined: 18-December 03
Member No.: 10538



QUOTE (instaud @ Jul 17 2009, 08:56) *
He says he is an audiophile, and that's the reason his answer is yes. I come to this conclusion because he has "experienced hearing loss ... from riding sports motorcycles", listens to cassettes and LPs, but on the other hand was not happy with the sound of his CD transport/DAC combination, so he needed a "jitter buster" appliance. So the big picture for me is biased research which is funded by those having an interest in a certain result.


Wait, Dr. Kunchur funded his own research?? Nice for those who can!

Actually the audio-related research reported in his three journal publications appears to have been funded (well, "partially') intramurally -- i.e., by his own university. I don't see any indication of industry bias. This is in contrast to the vast evil CONSPIRACY involving the AES, the low-end audio industry, Hydrogenaudio, the Freemasons, and our secret alien lizard overlords, as uncovered by certain brave Stereophile forum posters! laugh.gif

Go to the top of the page
+Quote Post
Axon
post Jul 17 2009, 18:30
Post #171





Group: Members (Donating)
Posts: 1985
Joined: 4-January 04
From: Austin, TX
Member No.: 10933



QUOTE (Woodinville @ Jul 16 2009, 15:11) *
Specifically, he plays one pulse, and then two half-amplitude pulses separated by 5 microseconds, and listeners with unimpaired hearing in good settings can hear the difference. Given that this is a comb filter with its first zero at 100kHz, being able to hear the difference at 20kHz isn't too surprising, now. So, this shows a mechanism for which purely in-band (i.e. dc - 20kHz) signal can allow the distinction. Let's see.... Down about .45dB is just at the DL. In really good conditions, for direct, immediate comparison, with no level roving, etc. So, what did the doctor actually discover? Nothing that I can tell.
Um, no?

His experiments only involve pulses by implication - ie, that playing two time-misaligned sine waves is equivalent to one sine wave plus two rectangular pulses, or that a first-order-filtered square wave can be roughly treated as a square wave plus two exponentially decaying pulses. Neither of which actually involve pulses spaced 5us apart. Granted, everybody involved (including Dr. Kunchur) seem to be all too happy to reduce this down to the spaced pulses idea, but let's keep it real on what is actually in the papers.

Combing does occur with his speaker-alignment experiment but he goes to lengths to measure the headphone output and assert that the DL is not reached at any frequency, and his method appears more or less satisfactory to me (at least as far as this particular part of the procedure is concerned).

QUOTE
This, of course, after misdirection upon misdirection from the people 'defending' his work
Yeah, there's no excuse for that, but seriously, what did you expect out of a gaggle of audiophiles like that? It's not all that productive to drag sasaudio etal into a fight that really should just be between you and Kunchur.

QUOTE (Woodinville @ Jul 16 2009, 22:43) *
Oh, and the best joke of all this was that the intentionally ignorant over there think I was being rude. Oh, baby, they have no idea what RUDE is. Apparently Gauss, Fourier, Fletcher and Zwicker are also rude, in any case. It's just like creationists, they only accept confirmatory data.
The impression I got was that Fourier (at the least) was a gasbag of a reasonably high order. I would not call him a role model.

QUOTE (2Bdecided @ Jul 17 2009, 04:10) *
It would be fun to repeat the experiment, but hook a CD recorder into the loop, testing if (proving that!) the "effect" was still audible after passing through 44.1kHz / 16-bits. Hopefully that would be simple enough to show how silly this is.
Curiously enough Kunchur never does run this control (ie, run the tests with 7khz sine instead of 7khz square, and/or 7+21khz sines). I'm unsure how important this is.

QUOTE (krabapple @ Jul 17 2009, 09:03) *
QUOTE (instaud @ Jul 17 2009, 08:56) *
He says he is an audiophile, and that's the reason his answer is yes. I come to this conclusion because he has "experienced hearing loss ... from riding sports motorcycles", listens to cassettes and LPs, but on the other hand was not happy with the sound of his CD transport/DAC combination, so he needed a "jitter buster" appliance. So the big picture for me is biased research which is funded by those having an interest in a certain result.


Wait, Dr. Kunchur funded his own research?? Nice for those who can!

Actually the audio-related research reported in his three journal publications appears to have been funded (well, "partially') intramurally -- i.e., by his own university. I don't see any indication of industry bias. This is in contrast to the vast evil CONSPIRACY involving the AES, the low-end audio industry, Hydrogenaudio, the Freemasons, and our secret alien lizard overlords, as uncovered by certain brave Stereophile forum posters! laugh.gif
Also note that Hi-Fi Critic did a feature on Kunchur (as an example of one of the "good guys", meaning somebody who justifies/apologizes for high end audio).

There's an extremely critical point to be made related to all of this, but I'm saving it to address to the good doctor himself.

This post has been edited by Axon: Jul 17 2009, 19:21
Go to the top of the page
+Quote Post
Ethan Winer
post Jul 17 2009, 19:04
Post #172





Group: Members
Posts: 248
Joined: 12-May 09
From: New Milford, CT
Member No.: 69730



QUOTE (instaud @ Jul 17 2009, 08:56) *
Prof. Kunchur ... listens to cassettes and LPs, but on the other hand was not happy with the sound of his CD transport/DAC combination, so he needed a "jitter buster" appliance.

That says an awful lot IMO. blink.gif

I wonder if he has any bass traps and other room treatment. laugh.gif

--Ethan

This post has been edited by Ethan Winer: Jul 17 2009, 19:06


--------------------
I believe in Truth, Justice, and the Scientific Method
Go to the top of the page
+Quote Post

7 Pages V  « < 5 6 7
Reply to this topicStart new topic
1 User(s) are reading this topic (1 Guests and 0 Anonymous Users)
0 Members:

 



RSS Lo-Fi Version Time is now: 25th December 2014 - 21:02